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User's Manual
Version 5.6
Document #: LTRT-68808
November 2008
SIP User's Manual
Contents
Table of Contents
1 Overview ........................................................................................................... 15 1.1 SIP Overview ..........................................................................................................16 2 Configuration Concepts ................................................................................... 17 3 Web-Based Management ................................................................................. 19 3.1 Computer Requirements ........................................................................................19 3.2 Accessing the Web Interface ..................................................................................20 3.3 Getting Acquainted with the Web Interface ............................................................21 3.3.1 Toolbar .................................................................................................................... 22 3.3.2 Navigation Tree ....................................................................................................... 23 3.3.2.1 Displaying Navigation Tree in Basic and Full View ................................. 24 3.3.2.2 Showing / Hiding the Navigation Pane .................................................... 25 3.3.3 Working with Configuration Pages .......................................................................... 25 3.3.3.1 Accessing Pages ..................................................................................... 26 3.3.3.2 Viewing Parameters ................................................................................ 26 3.3.3.3 Modifying and Saving Parameters........................................................... 28 3.3.3.4 Entering Phone Numbers in Various Tables ........................................... 30 3.3.3.5 Working with Tables ................................................................................ 30 3.3.4 Searching for Configuration Parameters ................................................................ 32 3.3.5 Working with Scenarios .......................................................................................... 34 3.3.5.1 Creating a Scenario ................................................................................. 34 3.3.5.2 Accessing a Scenario .............................................................................. 36 3.3.5.3 Editing a Scenario ................................................................................... 37 3.3.5.4 Saving a Scenario to a PC ...................................................................... 38 3.3.5.5 Loading a Scenario to the Device............................................................ 39 3.3.5.6 Deleting a Scenario ................................................................................. 40 3.3.5.7 Exiting Scenario Mode ............................................................................. 41 3.3.6 Customizing the Web Interface ............................................................................... 41 3.3.6.1 Replacing the Corporate Logo................................................................. 41 3.3.6.2 Customizing the Product Name ............................................................... 44 3.3.6.3 Creating a Login Welcome Message ...................................................... 44 3.3.7 Getting Help ............................................................................................................ 45 3.3.8 Using the Home Page ............................................................................................. 46 3.3.8.1 Assigning a Name to a Port ..................................................................... 47 3.3.8.2 Viewing Trunk Settings ............................................................................ 48 3.3.8.3 Switching Between Modules.................................................................... 48 3.3.9 Logging Off the Web Interface ................................................................................ 49 3.4 Configuration Tab ...................................................................................................50 3.4.1 Network Settings ..................................................................................................... 50 3.4.1.1 Configuring the IP Settings ...................................................................... 50 3.4.1.2 Configuring the Multiple Interface Table .................................................. 53 3.4.1.3 Configuring the Application Settings........................................................ 57 3.4.1.4 Configuring the NFS Settings .................................................................. 60 3.4.1.5 Configuring the IP Routing Table ............................................................ 62 3.4.1.6 Configuring the QoS Settings .................................................................. 63 3.4.2 Media Settings ........................................................................................................ 65 3.4.2.1 Configuring the Voice Settings ................................................................ 66 3.4.2.2 Configuring the Fax / Modem / CID Settings ........................................... 68 3.4.2.3 Configuring the RTP / RTCP Settings ..................................................... 72 3.4.2.4 Configuring the IPmedia Settings ............................................................ 76 3.4.2.5 Configuring the General Media Settings ................................................. 79 Version 5.6
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3.4.3 3.4.4 3.4.5 3.4.6 3.4.7 3.4.8 3.4.9 3.4.2.6 Configuring the DSP Templates .............................................................. 80 3.4.2.7 Configuring Media Security ..................................................................... 81 PSTN Settings......................................................................................................... 82 3.4.3.1 Configuring the Trunk Settings ................................................................ 82 3.4.3.2 Configuring the CAS State Machines ...................................................... 97 SS7 Configuration ................................................................................................... 99 Sigtran Configuration .............................................................................................. 99 Security Settings ..................................................................................................... 99 3.4.6.1 Configuring the Web User Accounts ....................................................... 99 3.4.6.2 Configuring the Web and Telnet Access List ........................................ 102 3.4.6.3 Configuring the Firewall Settings ........................................................... 103 3.4.6.4 Configuring the Certificates ................................................................... 105 3.4.6.5 Configuring the General Security Settings ............................................ 110 3.4.6.6 Configuring the IPSec Table.................................................................. 114 3.4.6.7 Configuring the IKE Table ..................................................................... 117 Protocol Configuration .......................................................................................... 120 3.4.7.1 Configuring the Protocol Definition Parameters .................................... 120 3.4.7.2 Configuring the SIP Advanced Parameters........................................... 151 3.4.7.3 Configuring the Number Manipulation Tables ....................................... 164 3.4.7.4 Configuring the Routing Tables ............................................................. 171 3.4.7.5 Configuring the Profile Definitions ......................................................... 190 3.4.7.6 Configuring the Trunk and IP Groups .................................................... 195 3.4.7.7 Configuring the Digital Gateway Parameters ........................................ 207 Advanced Applications .......................................................................................... 214 3.4.8.1 Configuring the Voice Mail (VM) Parameters ........................................ 214 3.4.8.2 Configuring RADIUS Accounting Parameters ....................................... 217 Configuring the TDM Bus Settings........................................................................ 218 3.5 Management Tab .................................................................................................220 3.5.1 Management Configuration................................................................................... 220 3.5.1.1 Configuring the Management Settings .................................................. 220 3.5.1.2 Configuring the Regional Settings ......................................................... 227 3.5.1.3 Maintenance Actions ............................................................................. 228 3.5.2 Software Update ................................................................................................... 231 3.5.2.1 Loading Auxiliary Files ........................................................................... 231 3.5.2.2 Upgrading the Software Upgrade Key ................................................... 233 3.5.2.3 Software Upgrade Wizard ..................................................................... 236 3.5.2.4 Backing Up and Restoring Configuration .............................................. 240 3.6 Status & Diagnostics Tab .....................................................................................241 3.6.1 Status & Diagnostics ............................................................................................. 242 3.6.1.1 Viewing the Device's Syslog Messages ................................................ 242 3.6.1.2 Viewing the Ethernet Port Information ................................................... 243 3.6.1.3 Viewing Active IP Interfaces .................................................................. 244 3.6.1.4 Viewing Device Information ................................................................... 244 3.6.1.5 Viewing Performance Statistics ............................................................. 245 3.6.1.6 Viewing Active Alarms ........................................................................... 246 3.6.1.7 Viewing Trunks & Channels Status ....................................................... 246 3.6.2 Gateway Statistics................................................................................................. 248 3.6.2.1 Call Counters ......................................................................................... 248 3.6.2.2 Call Routing Status ................................................................................ 250 3.6.2.3 SAS/SBC Registered Users .................................................................. 251 3.6.2.4 IP Connectivity ....................................................................................... 252 4 ini File Configuration ...................................................................................... 255 4.1 Secured Encoded ini File .....................................................................................255 4.2 The ini File Structure ............................................................................................256 4.2.1 Structure Rules ..................................................................................................... 256 4.2.2 Structure of Individual ini File Parameters ............................................................ 256 4.2.3 Structure of ini File Table Parameters .................................................................. 257 SIP User's Manual
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4.2.4 Contents
Example of an ini File ............................................................................................ 259 4.3 Modifying an ini File ..............................................................................................259 4.4 Reference for ini File Parameters .........................................................................260 4.4.1 Networking Parameters ........................................................................................ 260 4.4.2 System Parameters............................................................................................... 268 4.4.3 Web and Telnet Parameters ................................................................................. 274 4.4.4 Security Parameters.............................................................................................. 276 4.4.5 RADIUS Parameters ............................................................................................. 281 4.4.6 SNMP Parameters ................................................................................................ 282 4.4.7 SIP Configuration Parameters .............................................................................. 284 4.4.8 Media Server Parameters ..................................................................................... 301 4.4.9 Voice Mail Parameters .......................................................................................... 302 4.4.10 PSTN Parameters ................................................................................................. 303 4.4.11 ISDN and CAS Interworking-Related Parameters ................................................ 307 4.4.12 Number Manipulation and Routing Parameters.................................................... 314 4.4.13 Channel Parameters ............................................................................................. 325 4.4.14 Auxiliary / Configuration Files Parameters ............................................................ 331 5 Default Settings .............................................................................................. 333 5.1 Defining Default Settings ......................................................................................333 5.2 Restoring Factory Defaults ...................................................................................333 6 Auxiliary Configuration Files ......................................................................... 335 6.1 Configuring the Call Progress Tones File .............................................................335 6.2 Prerecorded Tones (PRT) File .............................................................................337 6.3 Voice Prompts File ...............................................................................................338 6.4 CAS Protocol Auxiliary Files .................................................................................339 6.5 Dial Plan File ........................................................................................................339 6.6 User Information File ............................................................................................340 7 IP Telephony Capabilities .............................................................................. 343 7.1 IP-to-IP Routing (SIP Trunking) ............................................................................343 7.2 Answer Machine Detector (AMD) .........................................................................343 7.3 Stand-Alone Survivability (SAS) Feature .............................................................346 7.3.1 Configuring SAS.................................................................................................... 347 7.3.2 Configuring Emergency Calls ............................................................................... 348 7.4 Configuring the DTMF Transport Types ...............................................................349 7.5 Fax and Modem Capabilities ................................................................................350 7.5.1 Fax/Modem Operating Modes .............................................................................. 350 7.5.2 Fax/Modem Transport Modes ............................................................................... 351 7.5.2.1 T.38 Fax Relay Mode ............................................................................ 351 7.5.2.2 Fax/Modem Bypass Mode ..................................................................... 352 7.5.2.3 Fax / Modem NSE Mode ....................................................................... 353 7.5.2.4 Fax / Modem Transparent Mode ........................................................... 354 7.5.2.5 Fax / Modem Transparent with Events Mode........................................ 355 7.5.2.6 G.711 Fax / Modem Transport Mode .................................................... 355 7.5.2.7 Fax Fallback .......................................................................................... 356 7.5.3 Supporting V.34 Faxes ......................................................................................... 356 7.5.3.1 Using Bypass Mechanism for V.34 Fax Transmission .......................... 356 7.5.3.2 Using Relay mode for both T.30 and V.34 faxes................................... 357 7.5.4 Supporting V.152 Implementation......................................................................... 357 Version 5.6
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7.6 Event Notification using X-Detect Header ............................................................358 7.7 RTP Multiplexing (ThroughPacket) ......................................................................360 7.8 Dynamic Jitter Buffer Operation ...........................................................................360 7.9 Configuring Alternative Routing (Based on Connectivity and QoS) ..................... 361 7.9.1 Alternative Routing Mechanism ............................................................................ 361 7.9.2 Determining the Availability of Destination IP Addresses ..................................... 362 7.9.3 PSTN Fallback as a Special Case of Alternative Routing .................................... 362 7.9.4 Relevant Parameters ............................................................................................ 362 7.10 Supported RADIUS Attributes ..............................................................................363 7.11 Call Detail Record ................................................................................................365 7.12 Trunk-to-Trunk Routing Example .........................................................................367 7.13 Proxy or Registrar Registration Example .............................................................368 7.14 Configuration Examples .......................................................................................369 7.14.1 SIP Call Flow......................................................................................................... 369 7.14.2 SIP Authentication Example ................................................................................. 371 7.14.3 SIP Trunking between Enterprise and ITSPs ....................................................... 373 7.15 Working with Supplementary Services .................................................................377 7.15.1 Call Hold and Retrieve .......................................................................................... 377 7.15.2 Call Transfer.......................................................................................................... 378 8 Networking Capabilities ................................................................................. 379 8.1 Ethernet Interface Configuration ..........................................................................379 8.2 Ethernet Interface Redundancy ............................................................................380 8.3 NAT (Network Address Translation) Support .......................................................380 8.3.1 STUN .................................................................................................................... 381 8.3.2 First Incoming Packet Mechanism ........................................................................ 382 8.3.3 No-Op Packets ...................................................................................................... 382 8.4 IP Multicasting ......................................................................................................383 8.5 Robust Reception of RTP Streams ......................................................................383 8.6 Multiple Routers Support ......................................................................................383 8.7 Simple Network Time Protocol Support ...............................................................383 8.8 IP QoS via Differentiated Services (DiffServ) .......................................................384 8.9 VLANS and Multiple IPs .......................................................................................384 8.9.1 Multiple IPs............................................................................................................ 384 8.9.2 IEEE 802.1p/Q (VLANs and Priority) .................................................................... 385 8.9.3 Getting Started with VLANS and Multiple IPs ....................................................... 387 8.9.3.1 Integrating Using the Web Interface ...................................................... 388 8.9.3.2 Integrating Using the ini File .................................................................. 390 9 Advanced PSTN Configuration ..................................................................... 393 9.1 Clock Settings ......................................................................................................393 9.2 Release Reason Mapping ....................................................................................394 9.2.1 Reason Header ..................................................................................................... 394 9.2.2 Fixed Mapping of ISDN Release Reason to SIP Response ................................. 394 9.2.3 Fixed Mapping of SIP Response to ISDN Release Reason ................................. 396 9.3 ISDN Overlap Dialing ...........................................................................................398 9.4 ISDN Non-Facility Associated Signaling (NFAS) .................................................398 9.4.1 NFAS Interface ID ................................................................................................. 399 9.4.2 Working with DMS-100 Switches .......................................................................... 400 SIP User's Manual
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9.4.3 Contents
Creating an NFAS-Related Trunk Configuration .................................................. 400 9.5 Redirect Number and Calling Name (Display) .....................................................401 9.6 Automatic Gain Control (AGC) .............................................................................402 10 Tunneling Applications .................................................................................. 403 10.1 TDM Tunneling .....................................................................................................403 10.2 QSIG Tunneling ....................................................................................................405 11 Supplied SIP Software Package .................................................................... 407 12 Selected Technical Specifications ................................................................ 409 13 Glossary .......................................................................................................... 415 Version 5.6
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List of Figures
Figure 1-1: Mediant 2000 Typical Application ....................................................................................... 16 Figure 3-1: Enter Network Password Screen ........................................................................................ 20 Figure 3-2: Main Areas of the Web Interface GUI ................................................................................. 21 Figure 3-3: "Reset" Displayed on Toolbar ............................................................................................. 22 Figure 3-4: Terminology for Navigation Tree Levels ............................................................................. 23 Figure 3-5: Navigation Tree in Basic and Full View ............................................................................... 24 Figure 3-6: Showing and Hiding Navigation Pane ................................................................................. 25 Figure 3-7: Toggling between Basic and Advanced Page View............................................................ 27 Figure 3-8: Expanding and Collapsing Parameter Groups .................................................................... 28 Figure 3-9: Editing Symbol after Modifying Parameter Value ............................................................... 29 Figure 3-10: Value Reverts to Previous Valid Value ............................................................................. 30 Figure 3-11: Adding an Index Entry to a Table ...................................................................................... 31 Figure 3-12: Compacting a Web Interface Table................................................................................... 32 Figure 3-13: Searched Result Screen ................................................................................................... 33 Figure 3-14: Scenario Creation Confirm Message Box ......................................................................... 34 Figure 3-15: Creating a Scenario........................................................................................................... 35 Figure 3-16: Scenario Loading Message Box ....................................................................................... 36 Figure 3-17: Scenario Example ............................................................................................................. 36 Figure 3-18: Scenario File Page ............................................................................................................ 38 Figure 3-19: Scenario Loading Message Box ....................................................................................... 40 Figure 3-20: Message Box for Confirming Scenario Deletion ............................................................... 40 Figure 3-21: Confirmation Message Box for Exiting Scenario Mode..................................................... 41 Figure 3-22: Customizing Web Logo and Product Name ...................................................................... 41 Figure 3-23: Image Download Screen ................................................................................................... 42 Figure 3-24: User-Defined Web Welcome Message after Login ........................................................... 44 Figure 3-25: Help Topic for Current Page ............................................................................................. 45 Figure 3-26: Areas of the Home Page ................................................................................................... 46 Figure 3-27: Shortcut Menu for Assigning a Port Name ........................................................................ 47 Figure 3-28: Entering the Port Name ..................................................................................................... 48 Figure 3-29: Click Module to which you want to Switch ........................................................................ 48 Figure 3-30: Confirmation Message Box for Switching Modules ........................................................... 49 Figure 3-31: Log Off Confirmation Box .................................................................................................. 49 Figure 3-32: Web Session Logged Off .................................................................................................. 49 Figure 3-33: IP Settings Page................................................................................................................ 51 Figure 3-34: Confirmation Message for Accessing the Multiple Interface Table ................................... 54 Figure 3-35: Interface Table Page ......................................................................................................... 54 Figure 3-36: Application Settings Page ................................................................................................. 57 Figure 3-37: NFS Settings Page ............................................................................................................ 60 Figure 3-38: IP Routing Table Page ..................................................................................................... 62 Figure 3-39: QoS Settings Page ............................................................................................................ 64 Figure 3-40: Voice Settings Page .......................................................................................................... 66 Figure 3-41: Fax/Modem/CID Settings Page ......................................................................................... 68 Figure 3-42: RTP / RTCP Settings Page ............................................................................................... 72 Figure 3-43: IPMedia Settings Page ...................................................................................................... 76 Figure 3-44: General Media Settings Page ........................................................................................... 79 Figure 3-45: DSP Templates Page ........................................................................................................ 80 Figure 3-46: Media Security Page ......................................................................................................... 81 Figure 3-47: Trunk Settings Page .......................................................................................................... 83 Figure 3-48: Trunk Scroll Bar ................................................................................................................. 84 Figure 3-49: CAS State Machine Page.................................................................................................. 97 Figure 3-50: Web User Accounts Page (for Users with 'Security Administrator' Privileges) ............... 100 Figure 3-51: Web & Telnet Access List Page - Add New Entry .......................................................... 102 Figure 3-52: Web & Telnet Access List Table ..................................................................................... 102 Figure 3-53: Firewall Settings Page..................................................................................................... 103 Figure 3-54: Certificates Signing Request Page ................................................................................. 106 Figure 3-55: IKE Table Listing Loaded Certificate Files ...................................................................... 107 Figure 3-56: General Security Settings Page ...................................................................................... 110 SIP User's Manual
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Figure 3-57: IPSec Table Page ........................................................................................................... 115 Figure 3-58: IKE Table Page ............................................................................................................... 118 Figure 3-59: SIP General Parameters Page ........................................................................................ 121 Figure 3-60: Proxy & Registration Page .............................................................................................. 133 Figure 3-61: Proxy Sets Table Page.................................................................................................... 141 Figure 3-62: Coders Page ................................................................................................................... 146 Figure 3-63: DTMF & Dialing Page...................................................................................................... 148 Figure 3-64: Advanced Parameters Page ........................................................................................... 152 Figure 3-65: Supplementary Services Page ........................................................................................ 159 Figure 3-66: SAS Configuration Page ................................................................................................. 162 Figure 3-67: SBC Settings Page.......................................................................................................... 164 Figure 3-68: Source Phone Number Manipulation Table for Tel-to-IP Calls ....................................... 165 Figure 3-69: Phone Context Table Page ............................................................................................. 170 Figure 3-70: Routing General Parameters Page ................................................................................. 172 Figure 3-71: Tel to IP Routing Page .................................................................................................... 176 Figure 3-72: Outbound IP Routing Page ............................................................................................. 179 Figure 3-73: IP to Trunk Group Routing Table Page ........................................................................... 182 Figure 3-74: Inbound IP Routing Table................................................................................................ 184 Figure 3-75: Internal DNS Table Page ................................................................................................ 186 Figure 3-76: Internal SRV Table Screen.............................................................................................. 187 Figure 3-77: Reasons for Alternative Routing Page ............................................................................ 188 Figure 3-78: Release Cause Mapping Page ....................................................................................... 189 Figure 3-79: Coder Group Settings Page ............................................................................................ 191 Figure 3-80: Tel Profile Settings Page ................................................................................................. 192 Figure 3-81: IP Profile Settings Page .................................................................................................. 194 Figure 3-82: Trunk Group Table Page ................................................................................................. 196 Figure 3-83: Trunk Group Settings Page ............................................................................................. 198 Figure 3-84: IP Group Table Page....................................................................................................... 201 Figure 3-85: Account Table Page ........................................................................................................ 205 Figure 3-86: Digital Gateway Parameters Page .................................................................................. 207 Figure 3-87: Voice Mail Settings Page ................................................................................................ 214 Figure 3-88: RADIUS Parameters Page .............................................................................................. 217 Figure 3-89: TDM Bus Settings Page .................................................................................................. 218 Figure 3-90: Management Settings Page ............................................................................................ 220 Figure 3-91: SNMP Trap Destinations Page ....................................................................................... 223 Figure 3-92: SNMP Community Strings Page ..................................................................................... 224 Figure 3-93: SNMP V3 Setting Page ................................................................................................... 225 Figure 3-94: SNMP Trusted Managers ................................................................................................ 227 Figure 3-95: Regional Settings Page ................................................................................................... 227 Figure 3-96: Maintenance Actions Page ............................................................................................. 228 Figure 3-97: Reset Confirmation Message Box................................................................................... 229 Figure 3-98: Device Lock Confirmation Message Box ........................................................................ 230 Figure 3-99: Load Auxiliary Files Page ................................................................................................ 232 Figure 3-100: Software Upgrade Key Status Page ............................................................................. 234 Figure 3-101: Software Upgrade Key with Multiple S/N Lines ............................................................ 235 Figure 3-102: Start Software Upgrade Wizard Screen ........................................................................ 237 Figure 3-103: Load CMP File Wizard Page ......................................................................................... 237 Figure 3-104: CMP File Successfully Loaded Message ...................................................................... 238 Figure 3-105: Load an ini File Wizard Page ........................................................................................ 239 Figure 3-106: End Process Wizard Page ............................................................................................ 240 Figure 3-107: Message Box Informing of Upgraded CMP File ............................................................ 240 Figure 3-108: Configuration File Page ................................................................................................. 241 Figure 3-109: Message Log Screen .................................................................................................... 242 Figure 3-110: Ethernet Port Information Page ..................................................................................... 243 Figure 3-111: Active IP Interfaces Page .............................................................................................. 244 Figure 3-112: Device Information Page ............................................................................................... 244 Figure 3-113: Performance Statistics Page ......................................................................................... 245 Figure 3-114: Active Alarms Page ....................................................................................................... 246 Version 5.6
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Figure 3-115: Trunks & Channels Status Page ................................................................................... 246 Figure 3-116: Example of a Selected Page Icon for Displaying Trunks 17-24 .................................... 247 Figure 3-117: Basic Channel Information Page................................................................................... 248 Figure 3-118: Calls Count Page .......................................................................................................... 249 Figure 3-119: Call Routing Status Page .............................................................................................. 251 Figure 3-120: SAS Registered Users Page ......................................................................................... 251 Figure 3-121: IP Connectivity Page ..................................................................................................... 252 Figure 6-1: Example of a User Information File ................................................................................... 341 Figure 7-1: Device's SAS Agent Redirecting Emergency Calls to PSTN ............................................ 348 Figure 7-2: SIP Call Flow ..................................................................................................................... 369 Figure 7-3: Example Setup for Routing Between ITSP and Enterprise PBX ...................................... 374 Figure 7-4: Configuring Proxy Set ID #1 in the Proxy Sets Table Page .............................................. 375 Figure 7-5: Configuring IP Groups #1 and #2 in the IP Group Table Page ......................................... 375 Figure 7-6: Assign the Trunk to Trunk Group ID #1 in the Trunk Group Table Page .......................... 376 Figure 7-7: Configuring Trunk Group #1 for Registration per Account in Trunk Group Settings Page 376 Figure 7-8: Configuring Accounts for PBX Registration to ITSPs in Account Table Page .................. 376 Figure 7-9: Configuring ITSP-to-Trunk Group #1 Routing in IP to Trunk Group Table Page.............. 376 Figure 7-10: Configuring Tel-to-IP Routing to ITSPs in Tel to IP Routing Table Page ....................... 376 Figure 8-1: NAT Architecture ............................................................................................................... 380 Figure 8-2: Multiple Network Interfaces and VLANs............................................................................ 386 Figure 8-3: VLAN Configuration in the IP Settings Page ..................................................................... 388 Figure 8-4: OAM, Control, Media IP Configuration in the IP Settings Page ........................................ 389 Figure 8-5: Multiple Interface Table Page ........................................................................................... 389 Figure 8-6: Static Routes for OAM/Control in IP Routing Table .......................................................... 390 SIP User's Manual
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Contents
List of Tables
Table 3-1: Description of Toolbar Buttons ............................................................................................. 22 Table 3-2: ini File Parameters for Changing Logo Image ...................................................................... 43 Table 3-3: ini File Parameters for Replacing Logo with Text ................................................................. 43 Table 3-4: ini File Parameters for Customizing Product Name ............................................................. 44 Table 3-5: ini File Parameter for Welcome Login Message................................................................... 44 Table 3-6: Description of the Areas of the Home Page ......................................................................... 46 Table 3-7: Network Settings -- IP Settings Parameters ......................................................................... 51 Table 3-8: Multiple Interface Table Parameters Description ................................................................. 55 Table 3-9: Application Settings Parameters .......................................................................................... 58 Table 3-10: Network Settings -- NFS Settings Parameters ................................................................... 61 Table 3-11: IP Routing Table Description .............................................................................................. 62 Table 3-12: QoS Settings Parameters ................................................................................................... 64 Table 3-13: Media Settings, Voice Settings Parameters ....................................................................... 66 Table 3-14: Media Settings -- Fax/Modem/CID Parameters ................................................................. 68 Table 3-15: Media Settings, RTP / RTCP Parameters .......................................................................... 73 Table 3-16: IPMedia Parameters ........................................................................................................... 77 Table 3-17: Media Settings Parameters ................................................................................................ 79 Table 3-18: DSP Templates Parameters ............................................................................................... 80 Table 3-19: Media Security Parameters ................................................................................................ 81 Table 3-20: Trunk (E1/T1/J1) Configuration Parameters ...................................................................... 85 Table 3-21: CAS State Machine Parameters Description ..................................................................... 98 Table 3-22: Web User Accounts Access Levels and Privileges ......................................................... 100 Table 3-23: Default Attributes for the Web User Accounts .................................................................. 100 Table 3-24: Internal Firewall Parameters............................................................................................. 104 Table 3-25: General Security Parameters ........................................................................................... 111 Table 3-26: Default IKE Second Phase Proposals .............................................................................. 115 Table 3-27: IPSec SPD Table Configuration Parameters ................................................................... 116 Table 3-28: Default IKE First Phase Proposals ................................................................................... 118 Table 3-29: IKE Table Configuration Parameters................................................................................ 119 Table 3-30: SIP General Parameters (Protocol Definition).................................................................. 122 Table 3-31: Proxy & Registration Parameters ..................................................................................... 134 Table 3-32: Proxy Sets Table Parameters .......................................................................................... 142 Table 3-33: Supported Coders ............................................................................................................ 145 Table 3-34: DTMF and Dialing Parameters ......................................................................................... 148 Table 3-35: Advanced Parameters Description ................................................................................... 153 Table 3-36: Supplementary Services Parameters ............................................................................... 159 Table 3-37: Stand-Alone Survivability Parameters Description ........................................................... 162 Table 3-38: SBC Parameters............................................................................................................... 164 Table 3-39: Number Manipulation Parameters Description................................................................. 166 Table 3-40: Dialing Plan Notations ...................................................................................................... 168 Table 3-41: NPI/TON Values for ISDN ETSI ....................................................................................... 169 Table 3-42: Phone-Context Parameters Description ........................................................................... 170 Table 3-43: Routing General Parameters Description......................................................................... 172 Table 3-44: Tel to IP Routing Table Parameters Description .............................................................. 177 Table 3-45: Outbound IP Routing Table Description ........................................................................... 179 Table 3-46: IP to Trunk Group Routing Table Description .................................................................. 183 Table 3-47: Inbound IP Routing Table Description.............................................................................. 185 Table 3-48: Description of Parameter Unique to IP Profile ................................................................. 195 Table 3-49: Trunk Group Table Description ........................................................................................ 196 Table 3-50: Trunk Group Settings Parameters Description................................................................. 198 Table 3-51: IP Group Parameters Description .................................................................................... 202 Table 3-52: Account Parameters Description ...................................................................................... 205 Table 3-53: Digital Gateway Parameters Description.......................................................................... 208 Table 3-54: Voice Mail Parameters ..................................................................................................... 215 Table 3-55: RADIUS Parameters Description ..................................................................................... 217 Table 3-56: TDM Bus Settings Parameters Description ...................................................................... 218 Version 5.6
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Table 3-57: Management Settings Parameters ................................................................................... 221 Table 3-58: SNMP Trap Destinations Parameters Description ........................................................... 223 Table 3-59: SNMP Community Strings Parameters Description ......................................................... 225 Table 3-60: SNMP V3 Users Parameters ............................................................................................ 226 Table 3-61: Auxiliary Files Descriptions............................................................................................... 231 Table 3-62: Ethernet Port Information Parameters .............................................................................. 243 Table 3-63: Color-Coding Icons for Trunk and Channel Status .......................................................... 247 Table 3-64: Call Counters Description ................................................................................................. 249 Table 3-65: Call Routing Status Parameters ....................................................................................... 251 Table 3-66: SAS Registered Users Parameters .................................................................................. 252 Table 3-67: IP Connectivity Parameters .............................................................................................. 253 Table 4-1: Networking ini File Parameters .......................................................................................... 260 Table 4-2: System ini File Parameters................................................................................................. 268 Table 4-3: Web and Telnet ini File Parameters ................................................................................... 274 Table 4-4: Security ini File Parameters................................................................................................ 276 Table 4-5: RADIUS ini File Parameters ............................................................................................... 281 Table 4-6: SNMP ini File Parameters .................................................................................................. 282 Table 4-7: SIP ini File Parameters ....................................................................................................... 284 Table 4-8: Media Server ini File Parameters ....................................................................................... 301 Table 4-9: Voice Mail ini File Parameters ............................................................................................ 302 Table 4-10: PSTN ini File Parameters ................................................................................................. 303 Table 4-11: ISDN and CAS Interworking-Related ini File Parameters ................................................ 307 Table 4-12: Number Manipulation and Routing ini File Parameters .................................................... 314 Table 4-13: Channel ini File Parameters ............................................................................................. 325 Table 4-14: Auxiliary / Configuration ini File Parameters .................................................................... 331 Table 6-1: User Information Items ....................................................................................................... 340 Table 7-1: Approximate AMD Detection Sensitivity (Based on North American English) ................... 344 Table 7-2: Supported X-Detect Event Types ....................................................................................... 358 Table 7-3: Supported RADIUS Attributes ............................................................................................ 363 Table 7-4: Supported CDR Fields........................................................................................................ 366 Table 8-1: Traffic / Network Types and Priority ................................................................................... 386 Table 8-2: Example of VLAN and Multiple IPs Configuration .............................................................. 387 Table 9-1: Mapping of ISDN Release Reason to SIP Response ........................................................ 394 Table 9-2: Mapping of SIP Response to ISDN Release Reason ........................................................ 396 Table 9-3: Calling Name (Display) ....................................................................................................... 401 Table 9-4: Redirect Number ................................................................................................................ 401 Table 11-1: Supplied Software Package ............................................................................................. 407 Table 12-1: Mediant 2000 Functional Specifications ........................................................................... 409 Table 13-1: Glossary of Terms ............................................................................................................ 415 SIP User's Manual
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Notices
Notice
This document describes the AudioCodes Mediant 2000 SIP gateway.
Information contained in this document is believed to be accurate and reliable at the time of
printing. However, due to ongoing product improvements and revisions, AudioCodes cannot
guarantee accuracy of printed material after the Date Published nor can it accept responsibility
for errors or omissions. Before consulting this document, check the corresponding Release
Notes regarding feature preconditions and/or specific support in this release. In cases where
there are discrepancies between this document and the Release Notes, the information in the
Release Notes supersedes that in this document. Updates to this document and other
documents can be viewed by registered customers at http://www.audiocodes.com/support.
© Copyright 2008 AudioCodes Ltd. All rights reserved.
This document is subject to change without notice.
Date Published: November-18-2008
Tip:
When viewing this manual on CD, Web site or on any other electronic copy,
all cross-references are hyperlinked. Click on the page or section numbers
(shown in blue) to reach the individual cross-referenced item directly. To
return back to the point from where you accessed the cross-reference, press
the ALT and Å keys
Trademarks
AC logo, Ardito, AudioCoded, AudioCodes, AudioCodes logo, CTI², CTI Squared, InTouch,
IPmedia, Mediant, MediaPack, MP-MLQ, NetCoder, Netrake, Nuera, Open Solutions
Network, OSN, Stretto, 3GX, TrunkPack, VoicePacketizer, VoIPerfect, What's Inside
Matters, Your Gateway To VoIP, are trademarks or registered trademarks of AudioCodes
Limited. All other products or trademarks are property of their respective owners.
WEEE EU Directive
Pursuant to the WEEE EU Directive, electronic and electrical waste must not be disposed
of with unsorted waste. Please contact your local recycling authority for disposal of this
product.
Customer Support
Customer technical support and service are provided by AudioCodes’ Distributors,
Partners, and Resellers from whom the product was purchased. For Customer support for
products purchased directly from AudioCodes, contact [email protected].
Abbreviations and Terminology
Each abbreviation, unless widely used, is spelled out in full when first used. Only industrystandard terms are used throughout this manual. Hexadecimal notation is indicated by 0x
preceding the number.
Version 5.6
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Related Documentation
Document #
Manual Name
LTRT-523xx (where xx is the
document version)
Product Reference Manual
LTRT-690xx
Mediant 3000 & Mediant 2000 & TP Series SIP Release Notes
LTRT-701xx
Mediant 2000 & IPmedia 2000 SIP-MGCP-MEGACO Installation
Manual
LTRT-665xx
CPE Configuration Guide for IP Voice Mail
LTRT-400xx
IP-to-IP SIP Call Routing Application Note
Warning: The device is supplied as a sealed unit and must only be serviced by
qualified service personnel.
Note: The term device, used throughout this manual, refers to the Mediant 2000
media gateway, unless otherwise specified.
Note: Where ‘network’ appears in this manual, it means Local Area Network (LAN),
Wide Area Network (WAN), etc. accessed via the device’s Ethernet interface.
Note: The terms IP-to-Tel and Tel-to-IP refer to the direction of the call relative to
the AudioCodes device. IP-to-Tel refers to calls received from the IP network
and destined to the PSTN/PBX (i.e., telephone connected directly or indirectly
to the device); Tel-to-IP refers to calls received from the PSTN/PBX and
destined for the IP network.
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1. Overview
Overview
This manual provides you with the information for installing, configuring, and operating the
Mediant 2000 SIP gateway (referred to throughout this manual as device).
The device is a SIP-based Voice-over-IP (VoIP) media gateway. the device enables voice,
fax, and data traffic to be sent over the same IP network.
The device provides excellent voice quality and optimized packet voice streaming over IP
networks. The device uses the award-winning, field-proven VoIPerfect™ voice compression
technology, typically implemented in AudioCodes products.
The device incorporates 1, 2, 4, 8 or 16 E1, T1, or J1 spans for direct connection to the
Public Switched Telephone Network (PSTN) / Private Branch Exchange (PBX) through
digital telephony trunks. The device also provides SIP trunking capabilities for Enterprises
operating with multiple Internet Telephony Service Providers (ITSP) for VoIP services. The
device includes two 10/100Base-TX Ethernet ports, providing redundancy connection to the
network.
The device supports up to 480 simultaneous VoIP or Fax over IP (FoIP) calls, supporting
various Integrated Services Digital Network (ISDN) Primary Rate Interface (PRI) protocols
such as EuroISDN, North American NI2, Lucent™ 4/5ESS, Nortel™ DMS100 and others. In
addition, it supports different variants of Channel Associated Signaling (CAS) protocols for
E1 and T1 spans, including MFC R2, E&M immediate start, E&M delay dial/start, loop start
and ground start.
The device, best suited for large and medium-sized VoIP applications is a compact device,
comprising a 19-inch, 1U chassis with optional dual AC or single DC power supplies. The
deployment architecture can include several devices in branch or departmental offices,
connected to local PBXs. Call routing is performed by the devices using internal routing or
SIP Proxy(s).
The device enables users to make cost-effective, long distance or international
telephone/fax calls between distributed company offices, using their existing
telephones/fax. These calls can be routed over the existing network using state-of-the-art
compression techniques, ensuring that voice traffic uses minimum bandwidth.
The device can also route calls over the network using SIP signaling protocol, enabling the
deployment of Voice over Packet solutions in environments where access is enabled to
PSTN subscribers by using a trunking device. This provides the ability to transmit voice and
telephony signals between a packet network and a TDM network.
Notes:
Version 5.6
•
The device is offered as a 1-module (up to 240 channels or 8 trunk
spans) or 2-module (for 480 channels or 16 trunk spans only) platform.
The latter configuration supports two TrunkPack modules, each having its
own IP address. Configuration instructions in this document relate to the
device as a 1-module platform and must be repeated for the second
module as well.
•
For channel capacity, refer to the device's specifications in ''Selected
Technical Specifications'' on page 409.
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The figure below illustrates a typical device applications VoIP network:
Figure 1-1: Mediant 2000 Typical Application
1.1
SIP Overview
Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol used on
the gateway for creating, modifying, and terminating sessions with one or more participants.
These sessions can include Internet telephone calls, media announcements, and
conferences.
SIP invitations are used to create sessions and carry session descriptions that enable
participants to agree on a set of compatible media types. SIP uses elements called Proxy
servers to help route requests to the user's current location, authenticate and authorize
users for services, implement provider call-routing policies and provide features to users.
SIP also provides a registration function that enables users to upload their current locations
for use by Proxy servers. SIP implemented in the gateway, complies with the Internet
Engineering Task Force (IETF) RFC 3261 (refer to http://www.ietf.org).
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2. Configuration Concepts
Configuration Concepts
You can configure the device's parameters (including upgrading the software, and
uploading configuration and auxiliary files), using the following tools:
„
An HTTP-based Embedded Web Server (Web interface), using any standard Web
browser (described in ''Web-based Management'' on page 19).
„
A configuration file referred to as the ini file (refer to ''ini File Configuration'' on page
255).
„
Simple Network Management Protocol (SNMP) browser software (refer to the Product
Reference Manual).
„
AudioCodes’ Element Management System (refer to AudioCodes’ EMS User’s Manual
or EMS Product Description).
Note: To initialize the device by assigning it an IP address, a firmware file (cmp),
and a configuration file (ini file), you can use AudioCodes' BootP/TFTP utility,
which accesses the device using its MAC address (refer to the Product
Reference Manual).
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Reader’s Notes
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3. Web-Based Management
Web-Based Management
The device's Embedded Web Server (Web interface) provides FCAPS (fault management,
configuration, accounting, performance, and security) functionality. The Web interface
allows you to remotely configure your device for quick-and-easy deployment, including
uploading of configuration (software upgrade) and auxiliary files, and resetting the device.
The Web interface provides real-time, online monitoring of the device, including display of
alarms and their severity. In addition, it displays performance statistics of voice calls and
related traffic parameters.
The Web interface provides a user-friendly, graphical user interface (GUI), which can be
accessed using any standard Web browser (e.g., Microsoft™ Internet Explorer). Access to
the Web interface is controlled by various security mechanisms such as login user name
and password, read / write privileges, and limiting access to specific IP addresses.
Notes:
3.1
•
The Web interface allows you to configure most of the device's
parameters. Those parameters that are not available in the Web interface
can be configured using the ini file.
•
Certain Web interface pages are feature-key dependant, and therefore,
only appear if your device's feature key supports the features relating to
these pages (refer to “Upgrading the Software Upgrade Key” on page
233).
•
Throughout this section, parameters enclosed in square brackets [...]
depict the ini file parameters for configuring the device using the ini file.
Computer Requirements
To use the device's Web interface, the following is required:
„
A connection to the Internet network (World Wide Web).
„
A network connection to the device's Web interface.
„
One of the following Web browsers:
„
•
Microsoft™ Internet Explorer™ (version 6.0 or later).
•
Netscape™ Navigator™ (version 7.2 or later).
•
Mozilla Firefox® (version 1.5.0.10 or later).
Recommended screen resolution of 1024 x 768 pixels, or 1280 x 1024 pixels.
Note: Your Web browser must be JavaScript-enabled in order to access the Web
interface.
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3.2
Accessing the Web Interface
The Web interface can be opened using any standard Web browser (refer to ''Computer
Requirements'' on page 19). When initially accessing the Web interface, use the default
user name ('Admin') and password ('Admin'). For changing the login user name and
password, refer to ''Configuring the Web User Accounts'' on page 99).
¾ To access the Web interface, take these 4 steps:
1.
Open a standard Web browser application.
2.
In the Web browser's Uniform Resource Locator (URL) field, specify the device's IP
address (e.g., http://10.1.10.10); the Web interface's 'Enter Network Password' dialog
box appears, as shown in the figure below:
Figure 3-1: Enter Network Password Screen
3.
In the 'User Name' and 'Password' fields, enter the case-sensitive, user name and
password.
4.
Click the OK button; the Web interface is accessed, displaying the 'Home' page (for a
detailed description of the 'Home' page, refer to ''Using the Home Page'' on page 46).
Note: If access to the device's Web interface is denied ("Unauthorized") due to
Microsoft Internet Explorer security settings, perform the following
troubleshooting procedures:
SIP User's Manual
1.
Delete all cookies in the Temporary Internet Files folder. If this does not
resolve the problem, the security settings may need to be altered
(continue with Step 2).
2.
In Internet Explorer, navigate to Tools menu > Internet Options >
Security tab > Custom Level, and then scroll down to the Logon options
and select Prompt for username and password. Select the Advanced
tab, and then scroll down until the HTTP 1.1 Settings are displayed and
verify that Use HTTP 1.1 is selected.
3.
Quit and start the Web browser again.
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3.3
3. Web-Based Management
Getting Acquainted with the Web Interface
The figure below displays the general layout of the Graphical User Interface (GUI) of the
Web interface:
Figure 3-2: Main Areas of the Web Interface GUI
The Web GUI is composed of the following main areas:
„
Title bar: Displays the corporate logo and product name. For replacing the logo with
another image or text, refer to ''Replacing the Corporate Logo'' on page 41. For
customizing the product name, refer to ''Customizing the Product Name'' on page 44.
„
Toolbar: Provides frequently required command buttons for configuration (refer to
''Toolbar'' on page 21).
„
Navigation Pane: Consists of the following areas:
„
Version 5.6
•
Navigation bar: Provides tabs for accessing the configuration menus (refer to
''Navigation Tree'' on page 23), creating a Scenario (refer to ''Scenarios'' on page
34), and searching ini file parameters that have corresponding Web interface
parameters (refer to ''Searching for Configuration Parameters'' on page 32).
•
Navigation tree: Displays the elements pertaining to the tab selected on the
Navigation bar (tree-like structure of the configuration menus, Scenario Steps, or
Search engine) .
Work pane: Displays configuration pages where all configuration is performed (refer to
''Working with Configuration Pages'' on page 25).
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3.3.1
Toolbar
The toolbar provides command buttons for quick-and-easy access to frequently required
commands, as described in the table below:
Table 3-1: Description of Toolbar Buttons
Icon
Button
Name
Submit
Description
Applies parameter settings to the device (refer to ''Saving
Configuration'' on page 230).
Note: This icon is grayed out when not applicable to the currently
opened page.
Saves parameter settings to flash memory (refer to ''Saving
Configuration'' on page 230).
Burn
Device
Actions
Opens a drop-down menu list with frequently needed commands:
ƒ
Load Configuration File: opens the 'Configuration File' page for
loading an ini file (refer to ''Backing Up and Restoring
Configuration'' on page 240).
ƒ
Save Configuration File: opens the 'Configuration File' page for
saving the ini file to a PC (refer to ''Backing Up and Restoring
Configuration'' on page 240).
ƒ
Reset: opens the 'Maintenance Actions' page for resetting the
device (refer to ''Resetting the Device'' on page 228).
ƒ
Software Upgrade Wizard: opens the 'Software Upgrade Wizard'
page for upgrading the device's software (refer to ''Software
Upgrade Wizard'' on page 236).
Home
Opens the 'Home' page (refer to ''Using the Home Page'' on page
46).
Help
Opens the Online Help topic of the currently opened configuration
page in the Work pane (refer to ''Getting Help'' on page 45).
Log off
Logs off a session with the Web interface (refer to ''Logging Off the
Web Interface'' on page 49).
Note: If you modify parameters that take effect only after a device reset, after you
click the Submit button, the toolbar displays the word "Reset" (in red color),
as shown in the figure below. This is a reminder to later save ('burn') your
settings to flash memory and reset the device.
Figure 3-3: "Reset" Displayed on Toolbar
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3.3.2
3. Web-Based Management
Navigation Tree
The Navigation tree, located in the Navigation pane, displays the menus (pertaining to the
menu tab selected on the Navigation bar) used for accessing the configuration pages. The
Navigation tree displays a tree-like structure of menus. You can easily drill-down to the
required page item level to open its corresponding page in the Work pane.
The terminology used throughout this manual for referring to the hierarchical structure of
the tree is as follows:
„
menu: first level (highest level)
„
submenu: second level - contained within a menu.
„
page item: last level (lowest level in a menu) - contained within a menu or submenu.
Figure 3-4: Terminology for Navigation Tree Levels
¾ To view menus in the Navigation tree, take this step:
„
Version 5.6
On the Navigation bar, select the required tab:
•
Configuration (refer to ''Configuration Tab'' on page 50)
•
Management (refer to ''Management Tab'' on page 220)
•
Status & Diagnostics (refer to ''Status & Diagnostics Tab'' on page 241)
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¾ To navigate to a page, take these 2 steps:
1.
2.
3.3.2.1
Navigate to the required page item, by performing the following:
•
Drilling-down using the plus
signs to expand the menus and submenus
•
Drilling-up using the minus
signs to collapse the menus and submenus
Select the required page item; the page opens in the Work pane.
Displaying Navigation Tree in Basic and Full View
You can view an expanded or reduced Navigation tree display regarding the number of
listed menus and submenus. This is relevant when using the configuration tabs
(Configuration, Management, and Status & Diagnostics) on the Navigation bar.
The Navigation tree menu can be displayed in one of two views:
„
Basic: displays only commonly used menus
„
Full: displays all the menus pertaining to a configuration tab.
The advantage of the Basic view is that it prevents "cluttering" the Navigation tree with
menus that may not be required. Therefore, a Basic view allows you to easily locate
required menus.
¾ To toggle between Full and Basic view, take this step:
„
Select the Basic option (located below the Navigation bar) to display a reduced menu
tree; select the Full option to display all the menus. By default, the Basic option is
selected.
Figure 3-5: Navigation Tree in Basic and Full View
Note: When in Scenario mode (refer to ''Scenarios'' on page 34), the Navigation tree
is displayed in 'Full' view (i.e., all menus are displayed in the Navigation tree).
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3. Web-Based Management
Showing / Hiding the Navigation Pane
The Navigation pane can be hidden to provide more space for elements displayed in the
Work pane. This is especially useful when the Work pane displays a page with a table that's
wider than the Work pane and to view the all the columns, you need to use scroll bars. The
arrow button located just below the Navigation bar is used to hide and show the Navigation
pane.
; the pane is hidden
„
To hide the Navigation pane: click the left-pointing arrow
and the button is replaced by the right-pointing arrow button.
„
; the pane is
To show the Navigation pane: click the right-pointing arrow
displayed and the button is replaced by the left-pointing arrow button.
Figure 3-6: Showing and Hiding Navigation Pane
3.3.3
Working with Configuration Pages
The configuration pages contain the parameters for configuring the device. The
configuration pages are displayed in the Work pane, which is located to the right of the
Navigation pane.
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3.3.3.1
Accessing Pages
The configuration pages are accessed by clicking the required page item in the Navigation
tree.
¾ To open a configuration page in the Work pane, take these 2 steps:
1.
On the Navigation bar, click the required tab:
•
Configuration (refer to ''Configuration Tab'' on page 50)
•
Management (refer to ''Management Tab'' on page 220)
•
Status & Diagnostics (refer to ''Status & Diagnostics Tab'' on page 241)
The menus of the selected tab appears in the Navigation tree.
2.
In the Navigation tree, drill-down to the required page item; the page opens in the
Work pane.
You can also access previously opened pages, by clicking your Web browser's Back button
until you have reached the required page. This is useful if you want to view pages in which
you have performed configurations in the current Web session.
Notes:
3.3.3.2
•
You can also access certain pages from the Device Actions button
located on the toolbar (refer to ''Toolbar'' on page 21).
•
To view all the menus in the Navigation tree, ensure that the Navigation
tree is in 'Full' view (refer to ''Displaying Navigation Tree in Basic and Full
View'' on page 24).
•
To get Online Help for the currently opened page, refer to ''Getting Help''
on page 45.
•
Certain pages may not be accessible if your Web user account's access
level is low (refer to ''Configuring the Web User Accounts'' on page 99).
Viewing Parameters
For convenience, some pages allow you to view a reduced or expanded display of
parameters. A reduced display allows you to easily identify required parameters, enabling
you to quickly configure your device.
The Web interface provides you with two methods for handling the display of page
parameters:
„
Display of "basic" and "advanced" parameters (refer to ''Displaying Basic and
Advanced Parameters'' on page 27)
„
Display of parameter groups (refer to ''Showing / Hiding Parameter Groups'' on page
28)
Note: Certain pages may only be read-only if your Web user account's access level
is low (refer to ''Configuring the Web User Accounts'' on page 99). If a page is
read-only, 'Read-Only Mode' is displayed at the bottom of the page.
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3.3.3.2.1 Displaying Basic and Advanced Parameters
Some pages provide you with an Advanced Parameter List / Basic Parameter List toggle
button that allows you to show or hide advanced parameters (in addition to displaying the
basic parameters). This button is located on the top-right corner of the page and has two
states:
„
Advanced Parameter List button with down-pointing arrow: click this button to display
all parameters.
„
Basic Parameter List button with up-pointing arrow: click this button to show only
common (basic) parameters.
The figure below shows an example of a page displaying basic parameters only, and then
showing advanced parameters as well, using the Advanced Parameter List button.
Figure 3-7: Toggling between Basic and Advanced Page View
For ease of identification, the basic parameters are displayed with a darker blue color
background than the advanced parameters.
Note: When the Navigation tree is in 'Full' mode (refer to ''Navigation Tree'' on page
23), configuration pages display all their parameters (i.e., the 'Advanced
Parameter List' view is displayed).
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3.3.3.2.2 Showing / Hiding Parameter Groups
Some pages provide groups of parameters, which can be hidden or shown. To toggle
between hiding and showing a group, simply click the group name button that appears
above each group. The button appears with a down-pointing or up-pointing arrow,
indicating that it can be collapsed or expanded when clicked, respectively.
Figure 3-8: Expanding and Collapsing Parameter Groups
3.3.3.3
Modifying and Saving Parameters
When you change parameter values on a page, the Edit
symbol appears to the right of
these parameters. This is especially useful for indicating the parameters that you have
currently modified (before applying the changes). After you save your parameter
modifications (refer to the procedure described below), the Edit symbols disappear.
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Figure 3-9: Editing Symbol after Modifying Parameter Value
¾ To save configuration changes on a page to the device's volatile
memory (RAM), take this step:
„
Click the Submit
button, which is located near the bottom of the page in which
you are working; modifications to parameters with on-the-fly capabilities are
immediately applied to the device and take effect; other parameters (displayed on the
symbol) are not changeable on-the-fly and require a device
page with the lightning
reset (refer to ''Resetting the Device'' on page 228) before taking effect.
Notes:
Version 5.6
•
Parameters saved to the volatile memory (by clicking Submit), revert to
their previous settings after a hardware or software reset (or if the device
is powered down). Therefore, to ensure parameter changes (whether onthe-fly or not) are retained, you need to save ('burn') them to the device's
non-volatile memory, i.e., flash (refer to ''Saving Configuration'' on page
230).
•
If you modify a parameter value and then attempt to navigate away from
the page without clicking Submit, a message box appears notifying you
of this. Click Yes to save your modifications or No to ignore them.
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If you enter an invalid parameter value (e.g., not in the range of permitted values) and then
click Submit, a message box appears notifying you of the invalid value. In addition, the
parameter value reverts to its previous value and is highlighted in red, as shown in the
figure below:
Figure 3-10: Value Reverts to Previous Valid Value
3.3.3.4
Entering Phone Numbers in Various Tables
Phone numbers or prefixes that you enter in various tables throughout the Web interface
such as the 'Tel to IP Routing' table, must only be entered as digits without any other
characters. For example, if you wish to enter the phone number 555-1212, it must be
entered as 5551212 without the hyphen (-). If the hyphen is entered, the entry is invalid.
3.3.3.5
Working with Tables
The Web interface includes many configuration pages that provide tables for configuring the
device. Some of these tables provide the following command buttons:
„
Add: adds an index entry to the table.
„
Duplicate: duplicates a selected, existing index entry.
„
Compact: organizes the index entries in ascending, consecutive order.
„
Delete: deletes a selected index entry.
„
Apply: saves the configuration.
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¾ To add an entry to a table, take these 2 steps:
1.
In the 'Add' field, enter the desired index entry number, and then click Add; an index
entry row appears in the table:
Figure 3-11: Adding an Index Entry to a Table
2.
Click Apply to save the index entry.
Notes:
•
Before you can add another index entry, you must ensure that you have
applied the previously added index entry (by clicking Apply).
•
If you leave the 'Add' field blank and then click Add, the existing index
entries are all incremented by one and the newly added index entry is
assigned the index 0.
¾ To add a copy of an existing index table entry, take these 3 steps:
1.
In the 'Index' column, select the index that you want to duplicate; the Edit button
appears.
2.
Click Edit; the fields in the corresponding index row become available.
3.
Click Duplicate; a new index entry is added with identical settings as the selected
index in Step 1. In addition, all existing index entries are incremented by one and the
newly added index entry is assigned the index 0.
¾ To edit an existing index table entry, take these 3 steps:
1.
In the 'Index' column, select the index corresponding to the table row that you want to
edit.
2.
Click Edit; the fields in the corresponding index row become available.
3.
Modify the values as required, and then click Apply; the new settings are applied.
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¾ To organize the index entries in ascending, consecutive order, take
the following step:
„
Click Compact; the index entries are organized in ascending, consecutive order,
starting from index 0. For example, if you added three index entries 0, 4, and 6, then
the index entry 4 is re-assigned index number 1 and the index entry 6 is re-assigned
index number 2.
Figure 3-12: Compacting a Web Interface Table
¾ To delete an existing index table entry, take these 3 steps:
3.3.4
1.
In the 'Index' column, select the index corresponding to the table row that you want to
delete.
2.
Click Delete; the table row is removed from the table.
Searching for Configuration Parameters
The Web interface provides a search engine that allows you to search any ini file parameter
that is configurable by the Web interface (i.e., has a corresponding Web parameter). You
can search for a specific parameter (e.g., "EnableIPSec") or a sub-string of that parameter
(e.g., "sec"). If you search for a sub-string, all parameters that contain the searched substring in their names are listed.
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¾ To search for ini file parameters configurable in the Web interface,
take these 4 steps:
1.
On the Navigation bar, click the Search tab; the Search engine appears in the
Navigation pane.
2.
In the 'Search' field, enter the parameter name or sub-string of the parameter name
that you want to search. If you have performed a previous search for such a
parameter, instead of entering the required string, you can use the 'Search History'
drop-down list to select the string (saved from a previous search).
3.
Click Search; a list of located parameters based on your search appears in the
Navigation pane.
Each searched result displays the following:
4.
•
ini file parameter name
•
Link (in green) to its location (page) in the Web interface
•
Brief description of the parameter
In the searched list, click the required parameter (link in green) to open the page in
which the parameter appears; the relevant page opens in the Work pane and the
searched parameter is highlighted for easy identification, as shown in the figure below:
Figure 3-13: Searched Result Screen
Note: If the searched parameter is not located, a notification message is displayed.
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3.3.5
Working with Scenarios
The Web interface allows you to create your own "menu" with up to 20 pages selected from
the menus in the Navigation tree (i.e., pertaining to the Configuration, Management, and
Status & Diagnostics tabs). The "menu" is a set of configuration pages grouped into a
logical entity referred to as a Scenario. Each page in the Scenario is referred to as a Step.
For each Step, you can select up to 25 parameters in the page that you want available in
the Scenario. Therefore, the Scenario feature is useful in that it allows you quick-and-easy
access to commonly used configuration parameters specific to your network environment.
When you login to the Web interface, your Scenario is displayed in the Navigation tree,
thereby, facilitating your configuration.
Instead of creating a Scenario, you can also load an existing Scenario from a PC to the
device (refer to ''Loading a Scenario to the Device'' on page 39).
3.3.5.1
Creating a Scenario
The Web interface allows you to create one Scenario with up to 20 configuration pages, as
described in the procedure below:
¾ To create a Scenario, take these 10 steps:
1.
On the Navigation bar, click the Scenarios tab; a message box appears, requesting
you to confirm creation of a Scenario:
Figure 3-14: Scenario Creation Confirm Message Box
Note: If a Scenario already exists, the Scenario Loading message box appears.
2.
Click OK; the Scenario mode appears in the Navigation tree as well as the menus of
the Configuration tab.
Note: If a Scenario already exists and you wish to create a new one, click the Create
Scenario button, and then click OK in the subsequent message box.
3.
In the 'Scenario Name' field, enter an arbitrary name for the Scenario.
4.
On the Navigation bar, click the Configuration or Management tab to display their
respective menus in the Navigation tree.
5.
In the Navigation tree, select the required page item for the Step, and then in the page
itself, select the required parameters by selecting the check boxes corresponding to
the parameters.
6.
In the 'Step Name' field, enter a name for the Step.
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Click the Next button located at the bottom of the page; the Step is added to the
Scenario and appears in the Scenario Step list:
Figure 3-15: Creating a Scenario
8.
Repeat steps 5 through 8 to add additional Steps (i.e., pages).
9.
When you have added all the required Steps for your Scenario, click the Save &
Finish button located at the bottom of the Navigation tree; a message box appears
informing you that the Scenario has been successfully created.
10. Click OK; the Scenario mode is quit and the menu tree of the Configuration tab
appears in the Navigation tree.
Notes:
Version 5.6
•
You can add up to 20 Steps to a Scenario, where each Step can contain
up to 25 parameters.
•
When in Scenario mode, the Navigation tree is in 'Full' display (i.e., all
menus are displayed in the Navigation tree) and the configuration pages
are in 'Advanced Parameter List' display (i.e., all parameters are shown in
the pages). This ensures accessibility to all parameters when creating a
Scenario. For a description on the Navigation tree views, refer to
''Navigation Tree'' on page 23.
•
If you previously created a Scenario and you click the Create Scenario
button, the previously created Scenario is deleted and replaced with the
one you are creating.
•
Only users with access level of 'Security Administrator' can create a
Scenario.
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3.3.5.2
Accessing a Scenario
Once you have created the Scenario, you can access it at anytime by following the
procedure below:
¾ To access the Scenario, take these 2 steps:
1.
On the Navigation bar, select the Scenario tab; a message box appears, requesting
you to confirm the loading of the Scenario.
Figure 3-16: Scenario Loading Message Box
2.
Click OK; the Scenario and its Steps appear in the Navigation tree, as shown in the
example figure below:
Figure 3-17: Scenario Example
When you select a Scenario Step, the corresponding page is displayed in the Work pane. In
each page, the available parameters are indicated by a dark-blue background; the
unavailable parameters are indicated by a gray or light-blue background.
To navigate between Scenario Steps, you can perform one of the following:
„
In the Navigation tree, click the required Scenario Step.
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In an opened Scenario Step (i.e., page appears in the Work pane), use the following
navigation buttons:
•
Next: opens the next Step listed in the Scenario.
•
Previous: opens the previous Step listed in the Scenario.
Note: If you reset the device while in Scenario mode, after the device resets, you
are returned once again to the Scenario mode.
3.3.5.3
Editing a Scenario
You can modify a Scenario anytime by adding or removing Steps (i.e., pages) or
parameters, and changing the Scenario name and the Steps' names.
Note: Only users with access level of 'Security Administrator' can edit a Scenario.
¾ To edit a Scenario, take these 6 steps:
1.
On the Navigation bar, click the Scenarios tab; a message box appears, requesting
you to confirm Scenario loading.
2.
Click OK; the Scenario appears with its Steps in the Navigation tree.
3.
Click the Edit Scenario button located at the bottom of the Navigation pane; the
'Scenario Name' and 'Step Name' fields appear.
4.
You can perform the following edit operations:
•
•
Version 5.6
Add Steps:
a.
On the Navigation bar, select the desired tab (i.e., Configuration or
Management); the tab's menu appears in the Navigation tree.
b.
In the Navigation tree, navigate to the desired page item; the corresponding
page opens in the Work pane.
c.
In the page, select the required parameter(s) by marking the corresponding
check box(es).
d.
Click Next.
Add or Remove Parameters:
a.
In the Navigation tree, select the required Step; the corresponding page
opens in the Work pane.
b.
To add parameters, select the check boxes corresponding to the desired
parameters; to remove parameters, clear the check boxes corresponding to
the parameters that you want removed.
c.
Click Next.
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•
•
•
3.3.5.4
Edit the Step Name:
a.
In the Navigation tree, select the required Step.
b.
In the 'Step Name' field, modify the Step name.
c.
In the page, click Next.
Edit the Scenario Name:
a.
In the 'Scenario Name' field, edit the Scenario name.
b.
In the displayed page, click Next.
Remove a Step:
a.
In the Navigation tree, select the required Step; the corresponding page
opens in the Work pane.
b.
In the page, clear all the check boxes corresponding to the parameters.
c.
Click Next.
5.
After clicking Next, a message box appears notifying you of the change. Click OK.
6.
Click Save & Finish; a message box appears informing you that the Scenario has
been successfully modified. The Scenario mode is exited and the menus of the
Configuration tab appear in the Navigation tree.
Saving a Scenario to a PC
You can save a Scenario to a PC (as a dat file). This is especially useful when requiring
more than one Scenario to represent different environment setups (e.g., where one
includes PBX interoperability and another not). Once you create a Scenario and save it to
your PC, you can then keep on saving modifications to it under different Scenario file
names. When you require a specific network environment setup, you can simply load the
suitable Scenario file from your PC (refer to ''Loading a Scenario to the Device'' on page
39).
¾ To save a Scenario to a PC, take these 5 steps:
1.
On the Navigation bar, click the Scenarios tab; the Scenario appears in the Navigation
tree.
2.
Click the Get/Send Scenario File button (located at the bottom of the Navigation tree);
the 'Scenario File' page appears, as shown below:
Figure 3-18: Scenario File Page
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3.
Click the Get Scenario File button; the 'File Download' window appears.
4.
Click Save, and then in the 'Save As' window navigate to the folder to where you want
to save the Scenario file. When the file is successfully downloaded to your PC, the
'Download Complete' window appears.
5.
Click Close to close the 'Download Complete' window.
Loading a Scenario to the Device
Instead of creating a Scenario, you can load a Scenario file (data file) from your PC to the
device.
¾ To load a Scenario to the device, take these 4 steps:
1.
On the Navigation bar, click the Scenarios tab; the Scenario appears in the Navigation
tree.
2.
Click the Get/Send Scenario File button (located at the bottom of the Navigation tree);
the 'Scenario File' page appears (refer to ''Saving a Scenario to a PC'' on page 38).
3.
Click the Browse button, and then navigate to the Scenario file stored on your PC.
4.
Click the Send File button.
Notes:
Version 5.6
•
You can only load a Scenario file to a device that has an identical
hardware configuration setup to the device in which it was created. For
example, if the Scenario was created in a device with FXS interfaces, the
Scenario cannot be loaded to a device that does not have FXS
interfaces.
•
The loaded Scenario replaces any existing Scenario.
•
You can also load a Scenario file using BootP, by loading an ini file that
contains the ini file parameter ScenarioFileName (refer to ''Web and
Telnet Parameters'' on page 273). The Scenario dat file must be located
in the same folder as the ini file. For a detailed description on BootP, refer
to the Product Reference Manual.
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3.3.5.6
Deleting a Scenario
You can delete the Scenario by using the Delete Scenario File button, as described in the
procedure below:
¾ To delete the Scenario, take these 4 steps:
1.
On the Navigation bar, click the Scenarios tab; a message box appears, requesting
you to confirm:
Figure 3-19: Scenario Loading Message Box
2.
Click OK; the Scenario mode appears in the Navigation tree.
3.
Click the Delete Scenario File button; a message box appears requesting
confirmation for deletion.
Figure 3-20: Message Box for Confirming Scenario Deletion
4.
Click OK; the Scenario is deleted and the Scenario mode closes.
Note: You can also delete a Scenario using the following alternative methods:
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•
Loading an empty dat file (refer to ''Loading a Scenario to the Device'' on
page 39).
•
Loading an ini file with the ScenarioFileName parameter set to no value
(i.e., ScenarioFileName = "").
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Exiting Scenario Mode
When you want to close the Scenario mode after using it for device configuration, follow the
procedure below:
¾ To close the Scenario mode, take these 2 steps:
1.
Simply click any tab (besides the Scenarios tab) on the Navigation bar, or click the
Cancel Scenarios button located at the bottom of the Navigation tree; a message box
appears, requesting you to confirm exiting Scenario mode, as shown below.
Figure 3-21: Confirmation Message Box for Exiting Scenario Mode
2.
3.3.6
Click OK to exit.
Customizing the Web Interface
You can customize the device's Web interface to suit your company preferences. The
following Web interface elements can be customized:
3.3.6.1
„
Corporate logo displayed on the Title bar (refer to ''Replacing the Corporate Logo'' on
page 41)
„
Product’s name displayed on the Title bar (refer to ''Customizing the Product Name'' on
page 44)
„
Login welcome message (refer to ''Creating a Login Welcome Message'' on page 44)
Replacing the Corporate Logo
The corporate logo that appears in the Title bar can be replaced either with a different logo
image (refer to ''Replacing the Corporate Logo with an Image'' on page 42) or text (refer to
''Replacing the Corporate Logo with Text'' on page 43).
The figure below shows an example of a customized Title bar. The top image displays the
Title bar with AudioCodes logo and product name. The bottom image displays a customized
Title bar with a different image logo and product name.
Figure 3-22: Customizing Web Logo and Product Name
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3.3.6.1.1 Replacing the Corporate Logo with an Image
You can replace the logo that appears in the Web interface's Title bar, using either the Web
interface or the ini file.
¾ To replace the default logo with a different image via the Web
interface, take these 7 steps:
1.
Access the device's Web interface (refer to ''Accessing the Web Interface'' on page
20).
2.
In the URL field, append the case-sensitive suffix ‘AdminPage’ to the IP address (e.g.,
http://10.1.229.17/AdminPage); the 'Admin' page appears.
3.
On the left pane, click Image Load to Device; the 'Image Download' page is
displayed, as shown in the figure below:
Figure 3-23: Image Download Screen
4.
Click the Browse button, and then navigate to the folder in which the logo image file
that you want to use is located.
5.
Click the Send File button; the image file uploads to the device. When loading is
complete, the page is automatically refreshed and the uploaded logo image is
displayed in the Web interface's title bar.
6.
If you want to modify the width of the image, in the 'Logo Width' field, enter the new
width (in pixels) and then click the Set Logo Width button.
7.
To save the image to flash memory, refer to ''Saving Configuration'' on page 230.
Notes:
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•
The logo image must be a GIF, JPG, or JPEG file.
•
The logo image must have a fixed height of 30 pixels. The width can be
up to 199 pixels, the default being 141 pixels.
•
The size of the image file can be up to 64 Kbytes.
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Tip:
If you encounter any problem during the loading of the file or you want to
restore the default image, click the Restore Default Images button.
¾ To replace the default logo with a different image using the ini file,
take these 3 steps:
1.
Place your corporate logo image file on the TFTP server in the same folder where the
device’s ini file is located.
2.
Configure the ini file parameters as described in the table below. (For a description on
using the ini file, refer to ''Modifying an ini File'' on page 259.)
3.
Load the ini file to the device using BootP / TFTP (i.e., not through the Web interface).
For detailed information on the BootP/TFTP application, refer to the Product Reference
Manual.
Table 3-2: ini File Parameters for Changing Logo Image
Parameter
Description
LogoFileName
The name of the image file for your corporate logo. Use a gif, jpg or jpeg
image file.
The default is AudioCodes’ logo file.
Note: The length of the name of the image file is limited to 48 characters.
LogoWidth
Width (in pixels) of the logo image.
The range is 0 - 199. The default value is 141 (which is the width of
AudioCodes’ displayed logo).
Note: The optimal setting depends on the screen resolution settings.
3.3.6.1.2 Replacing the Corporate Logo with Text
The corporate logo can be replaced with a text string instead of an image. To replace
AudioCodes’ default logo with a text string using the ini file, configure the ini file parameters
listed in the table below. (For a description on using the ini file, refer to ''Modifying an ini
File'' on page 259.)
Table 3-3: ini File Parameters for Replacing Logo with Text
Parameter
UseWebLogo
WebLogoText
Description
ƒ
[0] = Logo image is used (default).
ƒ
[1] = Text string used instead of a logo image.
Text string that replaces the logo image.
The string can be up to 15 characters.
Note: When a text string is used instead of a logo image, the Web browser’s title bar
displays the string assigned to the WebLogoText parameter.
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3.3.6.2
Customizing the Product Name
You can customize the product name (text) that appears in the Title bar, using the ini file
parameters listed in the table below. (For a description on using the ini file, refer to
''Modifying an ini File'' on page 259.)
Table 3-4: ini File Parameters for Customizing Product Name
Parameter
Description
UseProductName
UserProductName
3.3.6.3
Defines whether or not to change the product name:
ƒ
[0] = Don’t change the product name (default).
ƒ
[1] = Enable product name change.
The text string that replaces the product name.
The default is ‘Mediant 2000’.
The string can be up to 29 characters.
Creating a Login Welcome Message
You can create a Welcome message box (alert message) that appears after each successful login to
the device's Web interface. The ini file table parameter WelcomeMessage allows you to create the
Welcome message. Up to 20 lines of character strings can be defined for the message. If this
parameter is not configured, no Welcome message box is displayed after login.
An example of a Welcome message is shown in the figure below:
Figure 3-24: User-Defined Web Welcome Message after Login
Table 3-5: ini File Parameter for Welcome Login Message
Parameter
WelcomeMessage
Description
Defines the Welcome message that appears after a successful login to the
Web interface. The format of this parameter is as follows:
[WelcomeMessage]
FORMAT WelcomeMessage_Index = WelcomeMessage_Text;
[\WelcomeMessage]
For Example:
[WelcomeMessage ]
FORMAT WelcomeMessage_Index = WelcomeMessage_Text;
WelcomeMessage 1 = "*********************************";
WelcomeMessage 2 = "********* This is a Welcome message **";
WelcomeMessage 3 = "*********************************";
[\WelcomeMessage]
Note: Each index represents a line of text in the Welcome message box.
Up to 20 indices can be defined.
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Getting Help
The Web interface provides you with context-sensitive Online Help. The Online Help
provides you with brief descriptions of most of the parameters you'll need to successfully
configure the device. The Online Help provides descriptions of parameters pertaining to the
currently opened page.
¾ To view the Help topic for a currently opened page, take these 4
steps:
1.
Using the Navigation tree, open the required page for which you want Help.
2.
On the toolbar, click the Help
page appears, as shown below:
button; the Help topic pertaining to the opened
Figure 3-25: Help Topic for Current Page
3.
To view a description of a parameter, click the plus
sign.
To collapse the description, click the minus
4.
To close the Help topic, click the close
Help topic window.
sign to expand the parameter.
button located on the top-right corner of the
Note: Instead of clicking the Help button for each page you open, you can open it
once for a page, and then simply leave it open. Each time you open a
different page, the Help topic pertaining to that page is automatically
displayed.
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3.3.8
Using the Home Page
The 'Home' page provides you with a graphical display of the device's front panel,
displaying color-coded status icons for monitoring the functioning of the device. By default,
the 'Home' page is displayed when you access the device's Web interface. When you are
configuring the device (in a configuration page), you can always return to the 'Home' page,
by simply clicking the Home icon on the toolbar. The 'Home' page also displays general
device information (in the 'General Information' pane) such as the device's IP address and
firmware version.
¾ To access the Home page, take this step:
„
On the toolbar, click the Home
icon; the 'Home' page is displayed:
Figure 3-26: Areas of the Home Page
Note: The number of channels displayed in the 'Home' page depends on the
device's hardware configuration.
The table below describes the areas of the 'Home' page.
Table 3-6: Description of the Areas of the Home Page
Item# /
Label
1
Description
Displays the highest severity of an active alarm raised (if any) by the device:
ƒ
Green = No alarms
ƒ
Red = Critical alarm
ƒ
Orange = Major alarm
ƒ
Yellow = Minor alarm
You can also view a list of active alarms in the 'Active Alarms' page (refer to “Viewing
Active Alarms” on page 245), by clicking the Alarms area.
2
Blade Activity icon:
ƒ
3
(green): Initialization sequence terminated successfully.
Blade Fail icon:
ƒ
(gray): Normal functioning.
ƒ
(red): Blade failure.
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Item# /
Label
Description
T1/E1 Trunk Status icons for trunks 1 through 8.
4
ƒ
(gray): Disable - Trunk not configured (not in use).
ƒ
(green): Active OK - Trunk synchronized.
ƒ
(yellow): RAI Alarm - Remote Alarm Indication (RAI), also known as the 'Yellow'
Alarm.
ƒ
(red): LOS / LOF Alarm - Loss due to LOS (Loss of Signal) or LOF (Loss of
Frame).
ƒ
(blue): AIS Alarm - Alarm Indication Signal (AIS), also known as the 'Blue' Alarm
ƒ
(orange): D-Channel Alarm - D-channel alarm
You can switch modules (refer to “Switching Between Modules” on page 48), view port
settings (refer to “Viewing Trunk Settings” on page 48), and assign a name to a port
(refer to “Assigning a Name or Brief Description to a Port” on page 47).
Dual Ethernet Link icons:
5
ƒ
(gray): No link.
ƒ
(green): Active link.
You can also view detailed Ethernet port information in the 'Ethernet Port Information'
page (refer to “Viewing the Active Alarms Table” on page 245), by clicking this icon.
Dual Ethernet activity icons:
6
ƒ
ƒ
(gray): No Ethernet activity.
( orange): Transmit / receive activity.
7
T1/E1 Trunk Status icons for trunks 9 through 16. Refer to Item #4 for a description.
8
Power status icon:
(green): Power received by blade.
ƒ
(red): No power received by blade.
Slot status of installed blade in the chassis (SWAP Ready icon).
9
3.3.8.1
ƒ
Assigning a Name to a Port
The 'Home' page allows you to assign an arbitrary name or a brief description to each port.
This description appears as a tooltip when you move your mouse over the port.
¾ To add a port description, take these 3 steps:
1.
Click the required port icon; a shortcut menu appears, as shown below:
Figure 3-27: Shortcut Menu for Assigning a Port Name
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2.
From the shortcut menu, choose Update Port Info; a text box appears.
Figure 3-28: Entering the Port Name
3.
3.3.8.2
Type a brief description for the port, and then click Apply Port Info.
Viewing Trunk Settings
The 'Home' page allows you to view the settings of a selected port in the 'Trunk Settings'
screen. Accessing this screen from the Home page provides an alternative to accessing it
from the Advanced Configuration menu (refer to ''Configuring the Trunk Settings'' on page
82).
¾ To view port settings, take these 2 steps:
3.3.8.3
1.
On the 'Home' page, click a desired trunk port LED on the TP-1610 (labeled as items
#3 and #5 in the figure in Accessing the Home Page); a shortcut menu appears.
2.
From the shortcut menu, choose Port Settings; the 'Trunk Settings' screen opens.
Switching Between Modules
The device can house up to two modules, as discussed in previous sections. Since each
module is a standalone gateway, the 'Home' page displays only one of the modules to
which you are connected. However, you can easily switch to the second module, by having
the Web browser connect to the IP address of the other module.
¾ To switch modules, take these 3 steps:
1.
In the 'Home' page, click anywhere on the module to which you want to switch, as
shown below:
Figure 3-29: Click Module to which you want to Switch
A confirmation message box appears requesting you to confirm switching of modules.
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Figure 3-30: Confirmation Message Box for Switching Modules
3.3.9
2.
Click OK; the 'Enter Network Password' screen pertaining to the Web interface of the
switched module appears.
3.
Enter the login user name and password, and then click OK.
Logging Off the Web Interface
You can log off the Web interface and re-access it with a different user account. For
detailed information on the Web User Accounts, refer to User Accounts.
¾ To log off the Web interface, take these 2 steps:
1.
On the toolbar, click the Log Off
appears:
button; the 'Log Off' confirmation message box
Figure 3-31: Log Off Confirmation Box
2.
Click OK; the Web session is logged off and the Log In button appears.
Figure 3-32: Web Session Logged Off
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To log in again, simply click the Log In button, and then in the 'Enter Network Password'
dialog box, enter your user name and password (refer to ''Accessing the Web Interface'' on
page 20).
3.4
Configuration Tab
The Configuration tab on the Navigation bar displays all menus related to device
configuration. These menus appear in the Navigation tree and include the following:
„
Network Settings (refer to ''Network Settings'' on page 50)
„
Media Settings (refer to ''Media Settings'' on page 65)
„
PSTN Settings (refer to “PSTN Settings” on page 82)
„
SS7 Configuration (refer to “SS7 Configuration” on page 99)
„
Sigtran Configuration (refer to “Sigtran Configuration” on page 99)
„
Security Settings (refer to ''Security Settings'' on page 99)
„
Protocol Configuration (refer to ''Protocol Configuration'' on page 120)
„
Advanced Applications (refer to ''Advanced Applications'' on page 213)
„
TDM Configuration (refer to “Configuring the TDM Bus Settings” on page 218)
¾ To access the menus of the Configuration tab, take this step:
„
3.4.1
On the Navigation bar, click the Configuration tab; the Navigation tree displays the
configuration menus pertaining to the Configuration tab.
Network Settings
The Network Settings menu allows you to configure various networking parameters. This
menu contains the following page items:
3.4.1.1
„
IP Settings (refer to ''Configuring the IP Settings'' on page 50)
„
Application Settings (refer to ''Configuring the Application Settings'' on page 57)
„
IP Routing Table (refer to ''Configuring the IP Routing Table'' on page 62)
„
QoS Settings (refer to ''Configuring the QoS Settings'' on page 63)
Configuring the IP Settings
The 'IP Settings' page is used for configuring basic IP networking parameters such as the
device's IP address. However, from this page you can also access the 'Multiple Interface
Table' page for configuring multiple interfaces.
Note: Once you configure multiple interfaces in the 'Multiple Interface Table' page
(accessed by clicking the
button), when clicking the IP Settings page
item in the Navigation tree, the 'Multiple Interface Table' page is accessed
(instead of the 'IP Settings' page).
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¾ To configure the IP settings parameters, take these 4 steps:
1.
Open the 'IP Settings' page (Configuration tab > Network Settings menu > IP
Settings page item).
Figure 3-33: IP Settings Page
2.
Configure the IP parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-7: Network Settings -- IP Settings Parameters
Parameter
Description
IP Settings
IP Networking Mode
[EnableMultipleIPs]
Determines the IP network scheme.
ƒ
[0] Single IP Network = Single IP network (default).
ƒ
[1] Multiple IP Networks = Multiple IP networks (OAMP, Media,
and Control).
ƒ
[1] Dual IP (Media & Control) = Multiple IP networks.
ƒ
[1] Dual IP (OAM & Control) = Multiple IP networks.
ƒ
[1] Dual IP (OAM & Medial) = Multiple IP networks.
Note: This parameter is not relevant when using Multiple Interface
tables, activated by clicking the Multiple Interface Table button
described below (refer to ''Configuring the Multiple Interface Table'' on
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Parameter
Description
page 53). For detailed information on Multiple IPs, refer to ''Multiple
IPs'' on page 384.
Single IP Settings
IP Address
IP address of the device. Enter the IP address in dotted-decimal
notation, for example, 10.8.201.1.
Notes:
Subnet Mask
ƒ
A warning message is displayed (after clicking Submit) if the
entered value is incorrect.
ƒ
After changing the IP address, you must reset the device.
Subnet mask of the device. Enter the subnet mask in dotted-decimal
notation, for example, 255.255.0.0.
Notes:
Default Gateway Address
ƒ
A warning message is displayed (after clicking Submit) if the
entered value is incorrect.
ƒ
After changing the subnet mask, you must reset the device.
IP address of the default Gateway used by the device. Enter the IP
address in dotted-decimal notation, for example, 10.8.0.1.
Notes:
ƒ
A warning message is displayed (after clicking Submit) if the
entered value is incorrect.
ƒ
After changing the default Gateway IP address, you must reset the
device.
ƒ
For detailed information on multiple routers support, refer to
''Multiple Routers Support'' on page 383.
OAM Network Settings (Available only in Multiple IP and Dual IP modes.)
IP Address
[LocalOAMIPAddress]
The device's source IP address in the operations, administration,
maintenance, and provisioning (OAMP) network.
The default value is 0.0.0.0.
Subnet Mask
[LocalOAMSubnetMask]
The device's subnet mask in the OAMP network.
The default subnet mask is 0.0.0.0.
Default Gateway Address
[LocalOAMDefaultGW]
N/A. Use the IP Routing table instead (refer to ''Configuring the IP
Routing Table'' on page 62).
Control Network Settings (Available only in Multiple IP and Dual IP modes.)
IP Address
[LocalControlIPAddress]
The device's source IP address in the Control network.
The default value is 0.0.0.0.
Subnet Mask
[LocalControlSubnetMask]
The device's subnet mask in the Control network.
The default subnet mask is 0.0.0.0.
Default Gateway Address
[LocalControlDefaultGW]
N/A. Use the IP Routing table instead (refer to ''Configuring the IP
Routing Table'' on page 62).
Media Network Settings (Available only in Multiple IP and Dual IP modes.)
IP Address
[LocalMediaIPAddress]
The device's source IP address in the Media network.
The default value is 0.0.0.0.
Subnet Mask
[LocalMediaSubnetMask]
The device's subnet mask in the Media network.
The default subnet mask is 0.0.0.0.
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Parameter
Default Gateway Address
[LocalMediaDefaultGW]
Description
The device's default Gateway IP address in the Media network.
The default value is 0.0.0.0.
Multiple Interface Settings
Multiple Interface Table
button to open the 'Multiple Interface
Click the right-pointing arrow
Table' page. For a description of configuring multiple IP interfaces,
refer to ''Configuring the Multiple Interface Table'' on page 53.
VLAN (For detailed information on the device's VLAN implementation, refer to ''VLANS and Multiple
IPs'' on page 384.)
VLAN Mode
[VlANMode]
Enables the VLAN functionality.
ƒ
[0] Disable (default).
ƒ
[1] Enable.
Note: This parameter cannot be changed on-the-fly and requires a
device reset.
VALN ID Settings
Native VLAN ID
[VLANNativeVlanID]
Defines the native VLAN identifier (Port VLAN ID - PVID).
The valid range is 1 to 4094. The default value is 1.
OAM VLAN ID
[VLANOamVlanID]
Defines the OAMP VLAN identifier.
The valid range is 1 to 4094. The default value is 1.
Control VLAN ID
[VLANControlVlanID]
Defines the Control VLAN identifier.
The valid range is 1 to 4094. The default value is 2.
Media VLAN ID
[VLANMediaVlanID]
Defines the Media VLAN identifier.
The valid range is 1 to 4094. The default value is 3.
NAT Settings
NAT IP Address
[StaticNatIP]
3.4.1.2
Global (public) IP address of the device to enable static Network
Address Translation (NAT) between the device and the Internet.
Configuring the Multiple Interface Table
The 'Multiple Interface Table' page allows you to configure up to three logical network
interfaces, each with its own IP address, unique VLAN ID (if enabled), interface name, and
application types (i.e., Control, Media, and/or Operations, Administration, Maintenance and
Provisioning - OAMP) permitted on the interface. In addition, this page provides VLANrelated parameters for enabling VLANs, and for defining the 'Native' VLAN ID (VLAN ID to
which incoming, untagged packets are assigned). For assigning VLAN priorities and
Differentiated Services (DiffServ) for the supported Class of Service (CoS), refer to
''Configuring the QoS Settings'' on page 63.
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Notes:
•
Once you access the 'Multiple Interface Table' page, the 'IP Settings'
page is no longer available.
•
You can view all added IP interfaces that are currently active, in the 'IP
Active Interfaces' page (refer to ''Viewing Active IP Interfaces'' on page
244).
•
You can also configure this table using the ini file table parameter
InterfaceTable (refer to ''Networking Parameters'' on page 260).
¾ To configure the multiple IP interface table, take these 7 steps:
1.
Open the 'IP Settings' page (refer to ''Configuring the IP Settings'' on page 50).
2.
Under the Multiple Interface Settings group, click the right-arrow
Multiple Interface Table; a confirmation message box appears:
button alongside
Figure 3-34: Confirmation Message for Accessing the Multiple Interface Table
3.
Click OK to confirm; the 'Multiple Interface Table' page appears:
Figure 3-35: Interface Table Page
4.
In the 'Add' field, enter the desired index number for the new interface, and then click
Add; the index row is added to the table.
5.
Configure the interface according to the table below.
6.
Click the Apply button; the interface is immediately applied to the device.
7.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
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Notes:
•
When adding more than one interface to the table, ensure that you
enable VLANs, using the 'VLAN Mode' (VlANMode) parameter.
•
When booting using BootP/DHCP protocols (refer to the Product
Reference Manual), an IP address is obtained from the server. This
address is used as the OAMP address for this session, overriding the IP
address you configured in the 'Multiple Interface Table' page. The
address specified in this table takes effect only after you save the
configuration to the device's flash memory. This enables the device to
use a temporary IP address for initial management and configuration,
while retaining the address (defined in this table) for deployment.
•
For a detailed description on multiple IP interfaces and VLANs, refer to
''VLANS and Multiple IPs'' on page 384.
•
For a description of the Web interface's table command buttons (e.g.,
Duplicate and Delete), refer to ''Working with Tables'' on page 30.
Table 3-8: Multiple Interface Table Parameters Description
Parameter
Description
Table parameters
Index
Index of each interface.
The range is 0-3.
Note: Each interface index must be unique.
Types of applications that are allowed on the specific interface.
Application Type
ƒ
[0] OAM = Only Operations, Administration, Maintenance and
Provisioning (OAMP) applications (e.g., Web, Telnet, SSH, and SNMP)
are allowed on the interface.
ƒ
[1] Media = Only Media (i.e., RTP streams of voice/video) is allowed on
the interface.
ƒ
[2] Control = Only Call Control applications (e.g., SIP) are allowed on the
interface.
ƒ
[3] OAM & Media = Only OAMP and Media (RTP) applications are
allowed on the interface.
ƒ
[4] OAM & Control = Only OAMP and Call Control applications are
allowed on the interface.
ƒ
[5] Media & Control = Only Media (RTP) and Call Control applications are
allowed on the interface.
ƒ
[6] All = All the applications are allowed on the interface.
Notes:
IP Address
Version 5.6
ƒ
Only one IPv4 interface of OAM can be configured.
ƒ
Only one IPv4 interface of Control can be configured.
ƒ
At least one interface with Media must be configured.
The IPv4 IP address in dotted-decimal notation.
Note: Each interface must be assigned a unique IP address.
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Parameter
Prefix Length
Description
This column lists the number of ‘1’ bits in the subnet mask (i.e., replaces the
standard dotted-decimal representation of the subnet mask for IPv4
interfaces). For example: A subnet mask of 255.0.0.0 is represented by a
prefix length of 8 (i.e., 11111111 00000000 00000000 00000000), and a
subnet mask of 255.255.255.252 is represented by a prefix length of 30 (i.e.,
11111111 11111111 11111111 11111100).
The prefix length is a Classless Inter-Domain Routing (CIDR) style
presentation of a dotted-decimal subnet notation. The CIDR-style
presentation is the latest method for interpretation of IP addresses.
Specifically, instead of using eight-bit address blocks, it uses the variablelength subnet masking technique to allow allocation on arbitrary-length
prefixes (refer to http://en.wikipedia.org/wiki/Classless_InterDomain_Routing for more information).
The prefix length values range from 0 to 31.
Defines the IP address of the default gateway used by the device.
Notes:
ƒ
Only one default gateway can be configured for the device and it must be
configured on an interface for Media traffic. All other table entries for this
column must have the value 0.0.0.0.
ƒ
The default gateway's IP address must be in the same subnet as the
interface address.
ƒ
For configuring additional routing rules for other interfaces, refer to
''Configuring the IP Routing Table'' on page 62.
Gateway
VLAN ID
Interface Name
Defines the VLAN ID for each interface. When using VLANs, the VLAN ID
must be unique for each interface. Incoming traffic tagged with this VLAN ID
is routed to the corresponding interface, and outgoing traffic from that
interface is tagged with this VLAN ID.
Defines a string (up to 16 characters) to name this interface. This name is
displayed in management interfaces (Web, CLI and SNMP) for better
readability and has no functional use.
Note: The interface name is a mandatory parameter and must be unique for
each interface.
General Parameters
VLAN Mode
[VlANMode]
Native VLAN ID
[VLANNativeVlanID]
For a description of this parameter, refer to ''Configuring the IP Settings'' on
page 50.
Defines the VLAN ID to which untagged incoming traffic is assigned.
Outgoing packets sent to this VLAN are sent only with a priority tag (VLAN
ID = 0).
When this parameter is equal to one of the VLAN IDs in the Interface Table
(and VLANs are enabled), untagged incoming traffic is considered as an
incoming traffic for that interface. Outgoing traffic sent from this interface is
sent with the priority tag (tagged with VLAN ID = 0).
When this parameter is different from any value in the 'VLAN ID' column in
the Interface Table, untagged incoming traffic is discarded, and all outgoing
traffic is tagged.
Note: If this parameter is not set (i.e., default value is 1), but one of the
interfaces has a VLAN ID configured to 1, this interface is still considered the
‘Native’ VLAN. If you do not wish to have a ‘Native’ VLAN ID and want to use
VLAN ID 1, set this parameter to a value other than any VLAN ID in the
table.
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Configuring the Application Settings
The 'Application Settings' page is used for configuring various application parameters such
as Telnet.
¾ To configure the Application settings parameters, take these 4
steps:
1.
Open the 'Application Settings' page (Configuration tab > Network Settings menu >
Application Settings page item).
Figure 3-36: Application Settings Page
2.
Configure the Applications parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
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Table 3-9: Application Settings Parameters
Parameter
Description
NTP Settings (For detailed information on Network Time Protocol (NTP), refer to ''Simple Network
Time Protocol Support'' on page 383.)
NTP Server IP Address
[NTPServerIP]
IP address (in dotted-decimal notation) of the NTP server.
The default IP address is 0.0.0.0 (i.e., internal NTP client is
disabled).
NTP UTC Offset
[NTPServerUTCOffset]
Defines the Universal Time Coordinate (UTC) offset (in seconds)
from the NTP server.
The default offset is 0. The offset range is -43200 to 43200.
NTP Update Interval
[NTPUpdateInterval]
Defines the time interval (in seconds) that the NTP client requests
for a time update.
The default interval is 86400 (i.e., 24 hours). The range is 0 to
214783647.
Note: AudioCodes does not recommend setting this parameter to
beyond one month (i.e., 2592000 seconds).
Telnet Settings
Embedded Telnet Server
[TelnetServerEnable]
Enables or disables the device's embedded Telnet server. Telnet is
disabled by default for security reasons.
ƒ
[0] Disable (default)
ƒ
[1] Enable Unsecured
ƒ
[2] Enable Secured (SSL)
Note: Only the primary Web User Account (which has Security
Administration access level) can access the device using Telnet
(refer to ''Configuring the Web User Accounts'' on page 99).
Telnet Server TCP Port
[TelnetServerPort]
Defines the port number for the embedded Telnet server.
The valid range is all valid port numbers. The default port is 23.
Telnet Server Idle Timeout
[TelnetServerIdleDisconnect]
Defines the timeout (in minutes) for disconnection of an idle Telnet
session. When set to zero, idle sessions are not disconnected.
The valid range is any value. The default value is 0.
SSH Server Enable
[SSHServerEnable]
Enables or disables the embedded Secure Shell (SSH) server.
SSH Server Port
[SSHServerPort]
ƒ
[0] Disable (default)
ƒ
[1] Enable
Defines the port number for the embedded SSH server.
Range is any valid port number. The default port is 22.
DNS Settings
DNS Primary Server IP
[DNSPriServerIP]
IP address of the primary DNS server. Enter the IP address in
dotted-decimal notation, for example, 10.8.2.255.
Note: To use Fully Qualified Domain Names (FQDN) in the 'Tel to
IP Routing' table (or 'Outbound IP Routing' table if EnableSBC is
set to 1), you must define this parameter.
DNS Secondary Server IP
[DNSSecServerIP]
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IP address of the second DNS server. Enter the IP address in
dotted-decimal notation, for example, 10.8.2.255.
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Parameter
Description
STUN Settings
Enable STUN
[EnableSTUN]
Determines whether Simple Traversal of UDP through NATs
(STUN) is enabled.
ƒ
[0] Disable (default)
ƒ
[1] Enable
When enabled, the device functions as a STUN client and
communicates with a STUN server located in the public Internet.
STUN is used to discover whether the device is located behind a
NAT and the type of NAT. In addition, it is used to determine the IP
addresses and port numbers that the NAT assigns to outgoing
signaling messages (using SIP) and media streams (using RTP,
RTCP and T.38). STUN works with many existing NAT types and
does not require any special behavior from them. For detailed
information on STUN, refer to ''STUN'' on page 381.
Notes:
ƒ
For defining the STUN server domain name, use the ini file
parameter STUNServerDomainName (refer to ''Networking
Parameters'' on page 260).
ƒ
This parameter cannot be changed on-the-fly and requires a
device reset.
STUN Server Primary IP
[STUNServerPrimaryIP]
Defines the IP address of the primary STUN server.
The valid range is the legal IP addresses. The default value is
0.0.0.0.
STUN Server Secondary IP
[STUNServerSecondaryIP]
Defines the IP address of the secondary STUN server.
The valid range is the legal IP addresses. The default value is
0.0.0.0.
NFS Settings
NFS Table
For detailed information on configuring the NFS table, refer to
''Configuring the NFS Settings'' on page 60.
DHCP Settings
Enable DHCP
[DHCPEnable]
Determines whether Dynamic Host Control Protocol (DHCP) is
enabled.
ƒ
[0] Disable = Disable DHCP support on the device (default).
ƒ
[1] Enable = Enable DHCP support on the device.
After the device powers up, it attempts to communicate with a
BootP server. If a BootP server does not respond and if DHCP is
enabled, then the device attempts to obtain its IP address and other
networking parameters from the DHCP server.
Notes:
Version 5.6
ƒ
After you enable the DHCP server, perform the following
procedure:
1. Click the Submit button, and then save the configuration
(refer to ''Saving Configuration'' on page 230).
2. Perform a cold reset using the device's hardware reset button
(soft reset via Web interface doesn't trigger the BootP/DHCP
procedure and this parameter reverts to 'Disable').
ƒ
Throughout the DHCP procedure the BootP/TFTP application
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Parameter
Description
must be deactivated, otherwise, the device receives a response
from the BootP server instead of from the DHCP server.
3.4.1.4
ƒ
For additional information on DHCP, refer to the Product
Reference Manual.
ƒ
DHCPEnable is a special 'Hidden' parameter. Once defined and
saved in flash memory, its assigned value doesn't revert to its
default even if the parameter doesn't appear in the ini file.
Configuring the NFS Settings
Network File System (NFS) enables the device to access a remote server's shared files and
directories, and to handle them as if they're located locally. You can configure up to five
different NFS file systems. As a file system, the NFS is independent of machine types, OSs,
and network architectures. NFS is used by the device to load the cmp, ini, and auxiliary
files, using the Automatic Update mechanism (refer to Automatic Update Mechanism). Note
that an NFS file server can share multiple file systems. There must be a separate row for
each remote file system shared by the NFS file server that needs to be accessed by the
device.
¾ To add remote NFS file systems, take these 6 steps:
1.
Open the 'Application Settings' page (refer to ''Configuring the Application Settings'' on
page 57).
2.
Under the NFS Settings group, click the right-arrow
the 'NFS Settings' page appears.
button alongside NFS Table;
Figure 3-37: NFS Settings Page
3.
In the 'Add' field, enter the index number of the remote NFS file system, and then click
Add; an empty entry row appears in the table.
4.
Configure the NFS parameters according to the table below.
5.
Click the Apply button; the remote NFS file system is immediately applied, which can
be verified by the appearance of the 'NFS mount was successful' message in the
Syslog server.
6.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
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Notes:
•
To avoid terminating current calls, a row must not be deleted or modified
while the device is currently accessing files on that remote NFS file
system.
•
The combination of 'HostOrIP' and 'RootPath' must be unique for each
row in the table. For example, the table must include only one row with a
Host / IP of 192.168.1.1 and Root Path of /audio.
•
For a description of the web interface's table command buttons (e.g.,
Duplicate and Delete), refer to ''Working with Tables'' on page 30.
•
You can also configure the NFS table using the ini file table parameter
NFSServers (refer to ''Networking Parameters'' on page 260).
Table 3-10: Network Settings -- NFS Settings Parameters
Parameter
Description
Index
The row index of the remote file system.
The valid range is 0 to 4.
Host Or IP
The domain name or IP address of the NFS server. If a domain name is
provided, a DNS server must be configured.
Root Path
Path to the root of the remote file system in the format: /[path]. For
example, '/audio'.
NFS version used to access the remote file system.
NFS Version
ƒ
[2] NFS Version 2.
ƒ
[3] NFS Version 3 (default).
Authentication method used for accessing the remote file system.
Authentication Type
ƒ
[0] = Auth NULL.
ƒ
[1] = Auth UNIX (default).
UID
User ID used in authentication when using Auth UNIX.
The valid range is 0 to 65537. The default is 0.
GID
Group ID used in authentication when using Auth UNIX.
The valid range is 0 to 65537. The default is 1.
The VLAN type for accessing the remote file system.
VLAN Type
ƒ
[0] OAMP.
ƒ
[1] Media (default).
Note: This parameter applies only if VLANs are enabled or if Multiple
IPs is configured (refer to ''VLANS and Multiple IPs'' on page 384).
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3.4.1.5
Configuring the IP Routing Table
The 'IP Routing Table' page allows you to define up to 50 static IP routing rules for the
device. For example, you can define static routing rules for the OAMP and Control networks
since a default gateway is supported only for the Media traffic network (refer to ''Configuring
the Multiple Interface Table'' on page 53). Before sending an IP packet, the device
searches this table for an entry that matches the requested destination host / network. If
such an entry is found, the device sends the packet to the indicated router. If no explicit
entry is found, the packet is sent to the default gateway (configured in the 'IP Settings'
page-- refer to ''Configuring the IP Settings'' on page 50).
¾ To configure static IP routing, take these 3 steps:
1.
Open the 'IP Routing Table' page (Configuration tab > Network Settings menu > IP
Routing Table page item).
Figure 3-38: IP Routing Table Page
2.
In the 'Add a new table entry' group, add a new static routing rule according to the
parameters described in the table below.
3.
Click Add New Entry; the new routing rule is added to the IP routing table.
To delete a routing rule from the table, select the 'Delete Row' check box that corresponds
to the routing rule entry, and then click Delete Selected Entries.
Table 3-11: IP Routing Table Description
Parameter
Description
Destination IP Address
[RoutingTableDestinationsColumn]
Specifies the IP address of the destination host /
network.
Destination Mask
[RoutingTableDestinationMasksColumn]
Specifies the subnet mask of the destination host /
network.
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Parameter
Description
The address of the host / network you want to reach is determined by an AND operation that is
applied to the fields 'Destination IP Address' and 'Destination Mask'. For example, to reach the
network 10.8.x.x, enter 10.8.0.0 in the field 'Destination IP Address' and 255.255.0.0 in the field
'Destination Mask'. As a result of the AND operation, the value of the last two octets in the field
'Destination IP Address' is ignored.
To reach a specific host, enter its IP address in the field 'Destination IP Address' and 255.255.255.255
in the field 'Destination Mask'.
Gateway IP Address
[RoutingTableGatewaysColumn]
The IP address of the router (next hop) to which the
packets are sent if their destination matches the rules
in the adjacent columns.
Note: The Gateway address must be in the same
subnet on which the address is configured on the
'Multiple Interface Table' page (refer to ''Configuring the
Multiple Interface Table'' on page 53).
Metric
[RoutingTableHopsCountColumn]
The maximum number of allowed routers (hops)
between the device and destination.
Note: This parameter must be set to 1 for the routing
rule to be valid. Routing entries with Hop Count equals
0 are local routes set automatically by the device.
Interface
[RoutingTableInterfacesColumn]
Specifies the interface (network type) to which the
routing rule is applied.
ƒ
[0] = OAMP (default).
ƒ
[1] = Media.
ƒ
[2] = Control.
For detailed information on the network types, refer to
''Configuring the Multiple Interface Table'' on page 53.
3.4.1.6
Configuring the QoS Settings
The 'QoS Settings' page is used for configuring the Quality of Service (QoS) parameters.
This page allows you to assign VLAN priorities (IEEE 802.1p) and Differentiated Services
(DiffServ) for the supported Class of Service (CoS).
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¾ To configure QoS, take these 4 steps:
1.
Open the 'QoS Settings' page (Configuration tab > Network Settings menu > QoS
Settings page item).
Figure 3-39: QoS Settings Page
2.
Configure the QoS parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-12: QoS Settings Parameters
Parameter
Description
Priority Settings
Network Priority
[VLANNetworkServiceClassPriority]
Defines the priority for Network Class of Service
(CoS) content.
The valid range is 0 to 7. The default value is 7.
Media Premium Priority
[VLANPremiumServiceClassMediaPriority]
Defines the priority for the Premium CoS content
and media traffic.
The valid range is 0 to 7. The default value is 6.
Control Premium Priority
[VLANPremiumServiceClassControlPriority]
Defines the priority for the Premium CoS content
and control traffic.
The valid range is 0 to 7. The default value is 6.
Gold Priority
[VLANGoldServiceClassPriority]
Defines the priority for the Gold CoS content.
The valid range is 0 to 7. The default value is 4.
Bronze Priority
[VLANBronzeServiceClassPriority]
Defines the priority for the Bronze CoS content.
The valid range is 0 to 7. The default value is 2.
Differential Services (For detailed information on IP QoS using Differentiated Services, refer to ''IP
QoS via Differentiated Services (DiffServ)'' on page 384).
Network QoS
[NetworkServiceClassDiffServ]
Defines the DiffServ value for Network CoS
content.
The valid range is 0 to 63. The default value is 48.
Media Premium QoS
Defines the DiffServ value for Premium Media CoS
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Parameter
[PremiumServiceClassMediaDiffServ]
Description
content (only if IPDiffServ is not set in the selected
IP Profile).
The valid range is 0 to 63. The default value is 46.
Note: The value for the Premium Control DiffServ
is determined by the following (according to
priority):
Control Premium QoS
[PremiumServiceClassControlDiffServ]
ƒ
IPDiffServ value in the selected IP Profile.
ƒ
PremiumServiceClassMediaDiffServ.
Defines the DiffServ value for Premium Control
CoS content (only if ControlIPDiffserv is not set in
the selected IP Profile).
The valid range is 0 to 63. The default value is 40.
Note: The value for the Premium Control DiffServ
is determined by the following (according to
priority):
ƒ
ControlPDiffserv value in the selected IP Profile.
ƒ
PremiumServiceClassControlDiffServ.
Gold QoS
[GoldServiceClassDiffServ]
Defines the DiffServ value for the Gold CoS
content.
The valid range is 0 to 63. The default value is 26.
Bronze QoS
[BronzeServiceClassDiffServ]
Defines the DiffServ value for the Bronze CoS
content.
The valid range is 0 to 63. The default value is 10.
3.4.2
Media Settings
The Media Settings menu allows you to configure the device's channel parameters. These
parameters are applied to all the device's channels. This menu contains the following page
items:
„
Voice Settings (refer to ''Configuring the Voice Settings'' on page 66)
„
Fax/Modem/CID Settings (refer to ''Configuring the Fax / Modem / CID Settings'' on
page 67)
„
RTP/RTCP Settings (refer to ''Configuring the RTP / RTCP Settings'' on page 71)
„
IPmedia Settings (refer to “Configuring the IPmedia Settings” on page 76)
„
General Media Settings (refer to ''Configuring the General Media Settings'' on page 78)
„
DSP Templates (refer to “Configuring the DSP Templates” on page 79)
„
Media Security (refer to ''Configuring Media Security'' on page 80)
Notes:
Version 5.6
•
Channel parameters can be modified on-the-fly. Changes take effect from
the next call.
•
Some channel parameters can be configured per channel or call routing,
using profiles (refer to ''Configuring the Profile Definitions'' on page 190).
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3.4.2.1
Configuring the Voice Settings
The 'Voice Settings' page is used for configuring various voice parameters such as voice
volume.
¾ To configure the Voice parameters, take these 4 steps:
1.
Open the 'Voice Settings' page (Configuration tab > Media Settings menu > Voice
Settings page item).
Figure 3-40: Voice Settings Page
2.
Configure the Voice parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-13: Media Settings, Voice Settings Parameters
Parameter
Description
Voice Volume
[VoiceVolume]
Voice gain control (in decibels). This parameter sets the level for
the transmitted (IP-to-PSTN) signal.
The valid range is -32 to 31 dB. The default value is 0 dB.
Input Gain
[InputGain]
Pulse-code modulation (PCM) input gain control (in decibels). This
parameter sets the level for the received (PSTN-to-IP) signal.
The valid range is -32 to 31 dB. The default value is 0 dB.
Silence Suppression
[EnableSilenceCompression]
Silence Suppression is a method for conserving bandwidth on VoIP
calls by not sending packets when silence is detected.
ƒ
[0] Disable = Silence Suppression is disabled (default).
ƒ
[1] Enable = Silence Suppression is enabled.
ƒ
[2] Enable without Adaptation = A single silence packet is sent
during a silence period (applicable only to G.729).
Note: If the selected coder is G.729, the following rules determine
the value of the 'annexb' parameter of the fmtp attribute in the SDP:
SIP User's Manual
ƒ
If EnableSilenceCompression is 0: 'annexb=no'.
ƒ
If EnableSilenceCompression is 1: 'annexb=yes'.
ƒ
If EnableSilenceCompression is 2 and IsCiscoSCEMode is 0:
'annexb=yes'.
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Parameter
Description
ƒ
Echo Canceler
[EnableEchoCanceller]
If EnableSilenceCompression is 2 and IsCiscoSCEMode is 1:
'annexb=no'.
Determines whether echo cancellation is enabled and therefore,
echo from voice calls is removed.
ƒ
[0] Off = Echo Canceler is disabled.
ƒ
[1] On = Echo Canceler is enabled (default).
Note: This parameter is used to maintain backward compatibility.
DTMF Transport Type
[DTMFTransportType]
Determines the DTMF transport type.
ƒ
[0] DTMF Mute = Erases digits from voice stream and doesn't
relay to remote.
ƒ
[2] Transparent DTMF = Digits remain in voice stream.
ƒ
[3] RFC 2833 Relay DTMF = Erases digits from voice stream
and relays to remote according to RFC 2833 (default).
ƒ
[7] RFC 2833 Relay Rcv Mute = DTMFs are sent according to
RFC 2833 and muted when received.
Note: This parameter is automatically updated if one of the
following parameters is configured: TxDTMFOption or
RxDTMFOption.
MF Transport Type
[MFTransportType]
Not Applicable.
DTMF Volume (-31 to 0 dB)
[DTMFVolume]
DTMF gain control value (in decibels) to the TDM side.
The valid range is -31 to 0 dB. The default value is -11 dB.
CAS Transport Type
[CASTransportType]
Controls the ABCD signaling transport type over IP.
ƒ
[0] CAS Events Only = Disable CAS relay (default).
ƒ
[1] CAS RFC2833 Relay = Enable CAS relay mode using RFC
2833.
The CAS relay mode can be used with the TDM tunneling feature to
enable tunneling over IP for both voice and CAS signaling bearers.
DTMF Generation Twist
[DTMFGenerationTwist]
Version 5.6
Defines the range (in decibels) between the high and low frequency
components in the DTMF signal. Positive decibel values cause the
higher frequency component to be stronger than the lower one.
Negative values cause the opposite effect. For any parameter
value, both components change so that their average is constant.
The valid range is -10 to 10 dB. The default value is 0 dB.
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3.4.2.2
Configuring the Fax / Modem / CID Settings
The 'Fax/Modem/CID Settings' page is used for configuring fax, modem, and Caller ID
(CID) parameters.
¾ To configure the fax, modem, and CID parameters, take these 4
steps:
1.
Open the 'Fax/Modem/CID Settings' page (Configuration tab > Media Settings menu
> Fax/Modem/CID Settings page item).
Figure 3-41: Fax/Modem/CID Settings Page
2.
Configure the fax, Modem, and CID parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-14: Media Settings -- Fax/Modem/CID Parameters
Parameter
Fax Transport Mode
[FaxTransportMode]
Description
Fax transport mode used by the device.
ƒ
[0] Disable = transparent mode.
ƒ
[1] T.38 Relay = (default).
ƒ
[2] Bypass.
ƒ
[3] Events Only.
Note: This parameter is overridden by the parameter
IsFaxUsed (refer to ''SIP General Parameters'' on page
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Parameter
Description
121). If the parameter IsFaxUsed is set to 1 (T.38 Relay)
or 3 (Fax Fallback), then FaxTransportMode is always
set to 1 (T.38 relay).
Caller ID Transport Type
[CallerIDTransportType]
V.21 Modem Transport Type
[V21ModemTransportType]
V.22 Modem Transport Type
[V22ModemTransportType]
V.23 Modem Transport Type
[V23ModemTransportType]
V.32 Modem Transport Type
[V32ModemTransportType]
Determines the device's behavior for Caller ID detection.
ƒ
[0] Disable = Caller ID is not detected - DTMF digits
remain in the voice stream.
ƒ
[1] Relay = Caller ID is detected - DTMF digits are
erased from the voice stream.
ƒ
[3] Mute = Caller ID is detected - DTMF digits are
erased from the voice stream (default).
V.21 Modem Transport Type used by the device.
ƒ
[0] Disable = Disable (Transparent) -- default
ƒ
[1] Enable Relay = N/A
ƒ
[2] Enable Bypass.
ƒ
[3] Events Only = Transparent with Events.
V.22 Modem Transport Type used by the device.
ƒ
[0] Disable = Disable (Transparent)
ƒ
[1] Enable Relay = N/A
ƒ
[2] Enable Bypass = (default)
ƒ
[3] Events Only = Transparent with Events
V.23 Modem Transport Type used by the device.
ƒ
[0] Disable = Disable (Transparent)
ƒ
[1] Enable Relay = N/A
ƒ
[2] Enable Bypass = (default)
ƒ
[3] Events Only = Transparent with Events
V.32 Modem Transport Type used by the device.
ƒ
[0] Disable = Disable (Transparent)
ƒ
[1] Enable Relay = N/A
ƒ
[2] Enable Bypass = (default)
ƒ
[3] Events Only = Transparent with Events
Note: This option applies to V.32 and V.32bis modems.
V.34 Modem Transport Type
[V34ModemTransportType]
Version 5.6
V.90 / V.34 Modem Transport Type used by the device.
ƒ
[0] Disable = Disable (Transparent)
ƒ
[1] Enable Relay = N/A
ƒ
[2] Enable Bypass = (default)
ƒ
[3] Events Only = Transparent with Events
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Parameter
Fax Relay Redundancy Depth
[FaxRelayRedundancyDepth]
Description
Number of times that each fax relay payload is
retransmitted to the network.
ƒ
[0] = No redundancy (default).
ƒ
[1] = One packet redundancy.
ƒ
[2] = Two packet redundancy.
Note: This parameter is applicable only to non-V.21
packets.
Fax Relay Enhanced Redundancy Depth
[FaxRelayEnhancedRedundancyDepth]
Number of times that control packets are retransmitted
when using the T.38 standard.
The valid range is 0 to 4. The default value is 0.
Fax Relay ECM Enable
[FaxRelayECMEnable]
Determines whether the Error Correction Mode (ECM)
mode is used during fax relay.
Fax Relay Max Rate (bps)
[FaxRelayMaxRate]
ƒ
[0] Disable = ECM mode is not used during fax relay.
ƒ
[1] Enable = ECM mode is used during fax relay
(default).
Maximum rate (in bps), at which fax relay messages are
transmitted (outgoing calls).
ƒ
[0] 2400 = 2.4 kbps.
ƒ
[1] 4800 = 4.8 kbps.
ƒ
[2] 7200 = 7.2 kbps.
ƒ
[3] 9600 = 9.6 kbps.
ƒ
[4] 12000 = 12.0 kbps.
ƒ
[5] 14400 = 14.4 kbps (default).
Note: The rate is negotiated between the sides (i.e., the
device adapts to the capabilities of the remote side).
Fax/Modem Bypass Coder Type
[FaxModemBypassCoderType]
Coder used by the device when performing fax/modem
bypass. Usually, high-bit-rate coders such as G.711
should be used.
ƒ
[0] G.711Alaw= G.711 A-law 64 (default).
ƒ
[1] G.711Mulaw = G.711 μ-law.
Fax/Modem Bypass Packing Factor
[FaxModemBypassM]
Number of (20 msec) coder payloads that are used to
generate a fax/modem bypass packet.
The valid range is 1, 2, or 3 coder payloads. The default
value is 1 coder payload.
Fax Bypass Output Gain
[FaxBypassOutputGain]
Defines the fax bypass output gain control.
The range is -31 to +31 dB, in 1-dB steps. The default is
0 (i.e., no gain).
Modem Bypass Output Gain
[ModemBypassOutputGain]
Defines the modem bypass output gain control.
The range is -31 dB to +31 dB, in 1-dB steps. The
default is 0 (i.e., no gain).
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Parameter
Fax CNG Mode
[FaxCNGMode]
CNG Detector Mode
[CNGDetectorMode]
T.38 Max Datagram Size
[T38MaxDatagram]
Version 5.6
Description
Determines the device's behavior upon detection of a
CNG tone.
ƒ
[0] = Does not send a SIP Re-INVITE upon detection
of a fax CNG tone when CNGDetectorMode is set to
1 (default).
ƒ
[1] = Sends a SIP Re-INVITE upon detection of a fax
CNG tone when CNGDetectorMode is set to 1.
Determines whether the device detects the fax Calling
tone (CNG).
ƒ
[0] Disable = The originating device doesn’t detect
CNG; the CNG signal passes transparently to the
remote side (default).
ƒ
[1] Relay = CNG is detected on the originating side.
CNG packets are sent to the remote side according
to T.38 (if IsFaxUsed = 1) and the fax session is
started. A Re-INVITE message isn’t sent and the fax
session starts by the terminating device. This option
is useful, for example, when the originating device is
located behind a firewall that blocks incoming T.38
packets on ports that have not yet received T.38
packets from the internal network (i.e., originating
device). To also send a SIP Re-INVITE message
upon detection of a fax CNG tone in this mode, set
the parameter FaxCNGMode to 1.
ƒ
[2] Events Only = CNG is detected on the originating
side and a fax session is started by the originating
side using the Re-INVITE message. Usually, T.38 fax
session starts when the ‘preamble’ signal is detected
by the answering side. Some SIP devices don’t
support the detection of this fax signal on the
answering side and thus, in these cases it is possible
to configure the device to start the T.38 fax session
when the CNG tone is detected by the originating
side. However, this mode is not recommended.
Defines the maximum size of a T.38 datagram that the
device can receive. This value is included in the
outgoing SDP when T.38 is in use.
The valid range is 122 to 1,024. The default value is 122.
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3.4.2.3
Configuring the RTP / RTCP Settings
The 'RTP/RTCP Settings' page allows you to configure the Real-Time Transport Protocol
(RTP) and Real-Time Transport (RTP) Control Protocol (RTCP) parameters.
¾ To configure the RTP / RTCP parameters, take these 4 steps:
1.
Open the 'RTP/RTCP Settings' page (Configuration tab > Media Settings menu >
RTP / RTCP Settings page item).
Figure 3-42: RTP / RTCP Settings Page
2.
Configure the RTP / RTCP parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
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Table 3-15: Media Settings, RTP / RTCP Parameters
Parameter
Description
Dynamic Jitter Buffer Minimum
Delay
[DJBufMinDelay]
Minimum delay (in msec) for the Dynamic Jitter Buffer.
The valid range is 0 to 150. The default delay is 10.
Dynamic Jitter Buffer Optimization
Factor
[DJBufOptFactor]
Dynamic Jitter Buffer frame error / delay optimization factor.
The valid range is 0 to 13. The default factor is 10.
RTP Redundancy Depth
[RTPRedundancyDepth]
Note: For more information on Jitter Buffer, refer to ''Dynamic
Jitter Buffer Operation'' on page 360.
Notes:
ƒ
Set to 13 for data (fax and modem) calls.
ƒ
For more information on Jitter Buffer, refer to ''Dynamic
Jitter Buffer Operation'' on page 360.
Determines whether the device generates redundant packets.
ƒ
[0] 0 = Disable the generation of redundant packets
(default).
ƒ
[1] 1 = Enable the generation of RFC 2198 redundancy
packets.
Packing Factor
[RTPPackingFactor]
N/A. Controlled internally by the device according to the
selected coder.
Basic RTP Packet Interval
[BasicRTPPacketInterval]
N/A. Controlled internally by the device according to the
selected coder.
RTP Directional Control
[RTPDirectionControl]
N/A. Controlled internally by the device according to the
selected coder.
RFC 2833 TX Payload Type
[RFC2833TxPayloadType]
N/A. Use the ini file parameter RFC2833PayloadType instead.
RFC 2833 RX Payload Type
[RFC2833RxPayloadType]
N/A. Use the ini file parameter RFC2833PayloadType instead.
RFC 2198 Payload Type
[RFC2198PayloadType]
RTP redundancy packet payload type, according to RFC
2198.
The range is 96-127. The default is 104.
Note: This parameter is applicable only if RTP Redundancy
Depth = 1.
Fax Bypass Payload Type
[FaxBypassPayloadType]
Determines the fax bypass RTP dynamic payload type.
The valid range is 96 to 120. The default value is 102.
Enable RFC 3389 CN Payload Type
[EnableStandardSIDPayloadType]
Determines whether Silence Indicator (SID) packets are sent
according to RFC 3389.
Comfort Noise Generation
Negotiation
[ComfortNoiseNegotiation]
ƒ
[0] Disable = G.711 SID packets are sent in a proprietary
method (default).
ƒ
[1] Enable = SID (comfort noise) packets are sent with the
RTP SID payload type according to RFC 3389. Applicable
to G.711 and G.726 coders.
Enables negotiation and usage of Comfort Noise (CN).
ƒ
[0] Disable = Disable (default).
ƒ
[1] Enable = Enable.
The use of CN is indicated by including a payload type for CN
Version 5.6
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Parameter
Description
on the media description line of the SDP. The device can use
CN with a codec whose RTP timestamp clock rate is 8,000 Hz
(G.711/G.726). The static payload type 13 is used. The use of
CN is negotiated between sides. Therefore, if the remote side
doesn't support CN, it is not used.
Note: Silence Suppression must be enabled to generate CN.
RTP Base UDP Port
[BaseUDPPort]
Lower boundary of UDP port used for RTP, RTCP (RTP port +
1) and T.38 (RTP port + 2). The upper boundary is the Base
UDP Port + 10 * (number of device's channels).
The range of possible UDP ports is 6,000 to 64,000. The
default base UDP port is 6000.
For example: If the Base UDP Port is set to 6000 (default)
then:
1) The first channel uses the following ports RTP 6000, RTCP
6001, and T.38 6002, 2) the second channel uses RTP 6010,
RTCP 6011, and T.38 6012, etc.
Note: If RTP Base UDP Port is not a factor of 10, the following
message is generated: 'invalid local RTP port'.
For detailed information on the default RTP/RTCP/T.38 port
allocation, refer to the Product Reference Manual.
Remote RTP Base UDP Port
[RemoteBaseUDPPort]
Determines the lower boundary of UDP ports used for RTP,
RTCP and T.38 by a remote device. If this parameter is set to
a non-zero value, ThroughPacket™ (RTP multiplexing) is
enabled. The device uses this parameter (and BaseUDPPort)
to identify and distribute the payloads from the received
multiplexed IP packet to the relevant channels.
The valid range is the range of possible UDP ports: 6,000 to
64,000.
The default value is 0 (i.e., RTP multiplexing is disabled).
For detailed information on RTP multiplexing, refer to RTP
Multiplexing (ThroughPacket) on page 360.
Notes:
ƒ
The value of this parameter on the local device must equal
the value of BaseUDPPort on the remote device.
ƒ
To enable RTP multiplexing, the parameters
L1L1ComplexTxUDPPort and L1L1ComplexRxUDPPort
must be set to a non-zero value.
ƒ
When VLANs are implemented, RTP multiplexing is not
supported.
RTP Multiplexing Local UDP Port
[L1L1ComplexTxUDPPort]
Determines the local UDP port used for outgoing multiplexed
RTP packets (applies to RTP multiplexing).
The valid range is the range of possible UDP ports: 6,000 to
64,000.
The default value is 0 (i.e., RTP multiplexing is disabled).
This parameter cannot be changed on-the-fly and requires a
device reset.
RTP Multiplexing Remote UDP Port
[L1L1ComplexRxUDPPort]
Determines the remote UDP port to where the multiplexed
RTP packets are sent, and the local UDP port used for
incoming multiplexed RTP packets (applies to RTP
multiplexing).
The valid range is the range of possible UDP ports: 6,000 to
64,000.
The default value is 0 (i.e., RTP multiplexing is disabled).
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Parameter
Description
This parameter cannot be changed on-the-fly and requires a
device reset.
Note: All devices that participate in the same RTP multiplexing
session must use this same port.
RTCP XR Settings
(Note: For a detailed description of RTCP XR reports, refer to the Product Reference Manual.)
Enable RTCP XR
[VQMonEnable]
Enables voice quality monitoring and RTCP Extended Reports
(RTCP XR).
ƒ
[0] Disable = Disable (default)
ƒ
[1] Enable = Enables
Burst Threshold
[VQMonBurstHR]
Voice quality monitoring - excessive burst alert threshold. if set
to -1 (default), no alerts are issued.
Delay Threshold
[VQMonDelayTHR]
Voice quality monitoring - excessive delay alert threshold. if
set to -1 (default), no alerts are issued.
R-Value Delay Threshold
[VQMonEOCRValTHR]
Voice quality monitoring - end of call low quality alert
threshold. if set to -1 (default), no alerts are issued.
Minimum Gap Size
[VQMonGMin]
Voice quality monitoring - minimum gap size (number of
frames). The default is 16.
RTCP XR Report Mode
[RTCPXRReportMode]
Determines whether RTCP XR reports are sent to the Event
State Compositor (ESC), and if so, defines the interval in
which they are sent.
ƒ
[0] Disable = RTCP XR reports are not sent to the ESC
(default).
ƒ
[1] End Call = RTCP XR reports are sent to the ESC at the
end of each call.
ƒ
[2] End Call & Periodic = RTCP XR reports are sent to the
ESC at the end of each call and periodically according to
the parameter RTCPInterval.
RTCP XR Packet Interval
[RTCPInterval]
Defines the time interval (in msec) between adjacent RTCP
reports.
The interval range is 0 to 65,535. The default interval is 5,000.
Disable RTCP XR Interval
Randomization
[DisableRTCPRandomize]
Controls whether RTCP report intervals are randomized or
whether each report interval accords exactly to the parameter
RTCPInterval.
RTCP XR Collection Server
[RTCPXREscIP]
Version 5.6
ƒ
[0] Disable = Randomize (default)
ƒ
[1] Enable = No Randomize
IP address of the Event State Compositor (ESC). The device
sends RTCP XR reports to this server, using PUBLISH
messages. The address can be configured as a numerical IP
address or as a domain name.
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3.4.2.4
Configuring the IPmedia Settings
The 'IPMedia Settings' page allows you to configure the IP media parameters. This includes
Automatic Gain Control (AGC) parameters. AGC equalizes the energy of the output signal
to a required level. It estimates the energy of the incoming signal, calculates the essential
gain and performs amplification. Feedback ensures that the output signal is not clipped.
You can define the required Gain Slope in decibels/sec and the required energy threshold.
When the AGC first detects a signal in the input, it begins operating in Fast Mode. This
means that the Gain Slope is 8 dB/sec for the first 1.5 seconds. After this period, the Gain
Slope is changed to the user-defined value. You can disable or enable the feature by using
the ini file parameter AGCDisableFastAdaptation. After Fast Mode is used, the signal
should be off for two minutes in order to have the feature turned on again. (This feature is
designed so that AGC can fast-adapt when a conversation is started).
¾ To configure the IP media parameters, take these 4 steps:
1.
Open the 'IPMedia Settings' page (Configuration tab > Media Settings menu >
IPmedia Settings page item).
Figure 3-43: IPMedia Settings Page
2.
Configure the IP media parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
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Table 3-16: IPMedia Parameters
Parameter
Description
Enable Answer Detector
[EnableAnswerDetector]
N/A.
Answer Detector Activity Delay
[AnswerDetectorActivityDelay]
N/A.
Answer Detector Silence Time
[AnswerDetectorSilenceTime]
N/A.
Answer Detector Redirection
[AnswerDetectorRedirection]
N/A.
Answer Detector Sensitivity
[AnswerDetectorSensitivity]
Determines the Answer Detector sensitivity.
The range is 0 (most sensitive) to 2 (least sensitive). The default
is 0.
Answer Machine Detector
Sensitivity
[AMDDetectionSensitivity]
Determines the Answer Machine Detector (AMD) detection
sensitivity. AMD can be useful in automatic dialing applications. In
some of these applications, it is important to detect if a human
voice or an answering machine is answering the call. AMD can be
activated and de-activated only after a channel is already open.
The direction of the detection (PSTN or IP) can also be
configured.
The range is 0 to 7, where 0 is the best detection of an answering
machine and 7 is the best detection of a live call (i.e., voice
detected). The default is 3.
For a detailed description on AMD, refer to Answer Machine
Detector (AMD) on page 343.
Note: To enable the AMD feature, set the ini file parameter
EnableDSPIPMDetectors to 1.
Enable AGC
[EnableAGC]
Activates the Automatic Gain Control (AGC) mechanism. The
AGC mechanism adjusts the level of the received signal to
maintain a steady (configurable) volume level.
ƒ
[0] Disable (default).
ƒ
[1] Enable.
Note: For a description on AGC, refer to “Automatic Gain Control
(AGC)” on page 401.
AGC Slope
[AGCGainSlope]
Version 5.6
Determines the AGC convergence rate:
ƒ
0 = 0.25 dB/sec
ƒ
1 = 0.50 dB/sec
ƒ
2 = 0.75 dB/sec
ƒ
3 = 1.00 dB/sec (default)
ƒ
4 = 1.25 dB/sec
ƒ
5 = 1.50 dB/sec
ƒ
6 = 1.75 dB/sec
ƒ
7 = 2.00 dB/sec
ƒ
8 = 2.50 dB/sec
ƒ
9 = 3.00 dB/sec
ƒ
10 = 3.50 dB/sec
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Parameter
AGC Redirection
[AGCRedirection]
Description
ƒ
11 = 4.00 dB/sec
ƒ
12 = 4.50 dB/sec
ƒ
13 = 5.00 dB/sec
ƒ
14= 5.50 dB/sec
ƒ
15 = 6.00 dB/sec
ƒ
16 = 7.00 dB/sec
ƒ
17 = 8.00 dB/sec
ƒ
18 = 9.00 dB/sec
ƒ
19 = 10.00 dB/sec
ƒ
20 = 11.00 dB/sec
ƒ
21 = 12.00 dB/sec
ƒ
22 = 13.00 dB/sec
ƒ
23 = 14.00 dB/sec
ƒ
24 = 15.00 dB/sec
ƒ
25 = 20.00 dB/sec
ƒ
26 = 25.00 dB/sec
ƒ
27 = 30.00 dB/sec
ƒ
28 = 35.00 dB/sec
ƒ
29 = 40.00 dB/sec
ƒ
30 = 50.00 dB/sec
ƒ
31 = 70.00 dB/sec
Determines the AGC direction.
ƒ
[0] 0 = AGC works on signals from the TDM side (default).
ƒ
[1] 1 = AGC works on signals from the IP side.
AGC Target Energy
[AGCTargetEnergy]
Determines the signal energy value (dBm) that the AGC attempts
to attain.
The valid range is 0 to -63 dBm. The default value is -19 dBm.
Enable Energy Detector
[EnableEnergyDetector]
N/A
Energy Detector Quality Factor
[EnergyDetectorQualityFactor]
N/A
Energy Detector Threshold
[EnergyDetectorThreshold]
N/A
Enable Pattern Detector
[EnablePatternDetector]
Enables or disables the activation of the Pattern Detector (PD).
Valid options include:
SIP User's Manual
ƒ
[0] Disable = Disable (default)
ƒ
[1] Enable = Enable
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3. Web-Based Management
Configuring the General Media Settings
The 'General Media Settings' page allows you to configure various media parameters.
¾ To configure general media parameters, take these 4 steps:
1.
Open the 'General Media Settings' page (Configuration tab > Media Settings menu >
General Media Settings page item).
Figure 3-44: General Media Settings Page
2.
Configure the general media parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-17: Media Settings Parameters
Parameter
Max Echo Canceller Length
[MaxEchoCancellerLength]
Description
Determines the maximum Echo Canceler Length (in msec),
which is the maximum echo path delay (tail length) for which the
echo canceller is designed to operate:
ƒ
[0] Default = based on various internal device settings to
attain maximum channel capacity (default)
ƒ
[11] 64 msec
ƒ
[22] 128 msec
Notes:
Enable Continuity Tones
Version 5.6
ƒ
Using 128 msec reduces the channel capacity to 200
channels.
ƒ
Reset the device after modifying this parameter.
ƒ
It isn't necessary to configure the parameter
EchoCancellerLength as it automatically acquires its value
from this parameter.
N/A.
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3.4.2.6
Configuring the DSP Templates
The 'DSP Templates' page allows you to assign up to two DSP templates to the device. In
addition, you can define the percentage of DSP resources allocated per DSP template.
¾ To select DSP templates, take these 5 steps:
1.
Open the 'DSP Templates' page (Configuration tab > Media Settings menu > DSP
Templates page item).
Figure 3-45: DSP Templates Page
2.
Select an index row by clicking the corresponding 'Index' radio button.
3.
Click Edit, and then in the 'DSP Template Number' field, enter the desired DSP
template number.
4.
Click Apply to save your settings.
5.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Notes:
•
If you delete all the table entries, the device uses the default DSP
template.
•
For a description of the Web interface's table command buttons (e.g.,
Duplicate and Delete), refer to ''Working with Tables'' on page 30.
Table 3-18: DSP Templates Parameters
Parameter
Description
DSP Template Number
[DSPVersionTemplateNumber]
Determines the number of the DSP template load. Each load has
a different coder list, channel capacity, and supported features.
For the list of supported DSP template numbers (coders and
channel capacity), refer to the device's Release Notes. The
default is 0.
DSP Resources Percentage
Resource percentage used for the specified template.
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Configuring Media Security
The 'Media Security' page allows you to configure media security.
¾ To configure media security, take these 4 steps:
1.
Open the 'Media Security' page (Configuration tab > Media Settings menu > Media
Security page item).
Figure 3-46: Media Security Page
2.
Configure the media security parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-19: Media Security Parameters
Parameter
Media Security
[EnableMediaSecurity]
Media Security Behavior
[MediaSecurityBehaviour]
Disable Authentication On
Transmitted RTP Packets
[RTPAuthenticationDisableTx]
Version 5.6
Description
Enables Secure Real-Time Transport Protocol (SRTP).
ƒ
[0] Disable = SRTP is disabled (default).
ƒ
[1] Enable = SRTP is enabled.
Determines the device's mode of operation when SRTP is used
(EnableMediaSecurity = 1).
ƒ
[0] Preferable = The device initiates encrypted calls. If
negotiation of the cipher suite fails, an unencrypted call is
established. Incoming calls that don't include encryption
information are accepted.
ƒ
[1] Mandatory = The device initiates encrypted calls, but if
negotiation of the cipher suite fails, the call is terminated.
Incoming calls that don't include encryption information are
rejected (default).
On a secured RTP session, this parameter determines whether
to enable Authentication on transmitted RTP packets.
ƒ
[0] Enable (default)
ƒ
[1] Disable
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Parameter
Disable Encryption On
Transmitted RTP Packets
[RTPEncryptionDisableTx]
Disable Encryption On
Transmitted RTCP Packets
[RTCPEncryptionDisableTx]
Description
On a secured RTP session, this parameter determines whether
to enable Encryption on transmitted RTP packets.
ƒ
[0] Enable (default)
ƒ
[1] Disable
On a secured RTP session, this parameter determines whether
to enable Encryption on transmitted RTCP packets.
ƒ
[0] Enable (default)
ƒ
[1] Disable
SRTP Settings
Master Key Identifier (MKI) Size
[SRTPTxPacketMKISize]
3.4.3
Determines the size (in bytes) of the Master Key Identifier (MKI)
in SRTP Tx packets.
The range is 0 to 4. The default value is 0.
PSTN Settings
The PSTN Settings menu allows you to configure various PSTN settings and includes the
following page items:
3.4.3.1
„
Trunk Settings (refer to ''Configuring the Trunk Settings'' on page 82)
„
CAS State Machines (refer to ''Configuring the CAS State Machines'' on page 97)
Configuring the Trunk Settings
The 'Trunk Settings' page allows you to configure the device's trunks. For configuring the
trunks using the ini file parameters, refer to ''PSTN Parameters'' on page 303.
¾ To configure the Trunks, take these 7 steps:
1.
Open the ‘Trunk Settings’ page (Configuration tab > PSTN Settings menu > Trunk
Settings page item).
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Figure 3-47: Trunk Settings Page
Version 5.6
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On the top of the page, a bar with Trunk number icons displays the status of each
trunk, according to the following color codes:
2.
•
Grey: Disabled
•
Green: Active
•
Yellow: RAI alarm
•
Red: LOS / LOF alarm
•
Blue: AIS alarm
•
Orange: D-channel alarm (ISDN only)
Select the trunk that you want to configure, by clicking the desired Trunk number icon.
The bar initially displays the first eight trunk number icons (i.e., trunks 1 through 8). To
scroll through the trunk number icons (i.e., view the next/last or previous/first group of
eight trunks), refer to the figure below:
Figure 3-48: Trunk Scroll Bar
Note: If the Trunk scroll bar displays all the available trunks, the scroll bar buttons
are unavailable.
After you have selected a trunk, the following is displayed:
1.
•
The read-only 'Trunk ID' field displays the selected trunk number.
•
The read-only ‘Trunk Configuration State’ displays the state of the trunk (e.g.,
'Active' or 'Inactive').
•
The parameters displayed in the page pertain to the selected trunk only.
Click the Stop Trunk
button (located at the bottom of the page) to de-activate the
trunk so that you can configure currently grayed out (unavailable) parameters.(Skip
this step if you want to configure parameters that are also available when the trunk is
active). The stopped trunk is indicated by the following:
•
The ‘Trunk Configuration State’ field displays ‘Inactive’.
•
The Stop Trunk button is replaced by the Apply Trunk Settings
(When all trunks are stopped, the Apply to All Trunks
•
2.
button.
button also appears.)
All the parameters are available and can be modified.
Configure the desired trunk parameters, as described in the table below.
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3.
Click the Apply Trunk Settings button to apply the changes to the selected trunk (or
click Apply to All Trunks to apply the changes to all trunks); the Stop Trunk button
replaces Apply Trunk Settings and the ‘Trunk Configuration State’ displays 'Active'.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
5.
To reset the device, refer to ''Resetting the Device'' on page 228.
Notes:
•
If the ‘Protocol Type’ field displays 'NONE' (i.e., no protocol type
selected) and no other trunks have been configured, after selecting a PRI
protocol type, you must reset the device.
•
The displayed parameters on the page depend on the protocol selected
in the ‘Protocol Type’ field.
•
All trunks must be of the same line type (i.e., either E1 or T1). However,
different variants of the same line type can be configured on different
trunks, for example, E1 Euro ISDN and E1 CAS (subject to the
constraints in the device's Release Notes).
•
If the trunk protocol type is CAS, you can assign or modify a dial plan (in
the 'Dial Plan' field) and perform this without stopping the trunk.
•
If the trunk can’t be stopped because it provides the device’s clock
(assuming the device is synchronized with the E1/T1 clock), assign a
different E1/T1 trunk to provide the device’s clock or enable ‘TDM Bus
PSTN Auto Clock’ in the 'TDM Bus Settings' page (refer to ''Configuring
the TDM Bus Settings'' on page 218).
•
To delete a previously configured trunk, set the parameter 'Protocol Type'
to 'None'.
Table 3-20: Trunk (E1/T1/J1) Configuration Parameters
ini File Field Name
Web Parameter Name
Valid Range and Description
General Settings
Protocol Type
[ProtocolType]
Version 5.6
Defines the PSTN protocol for the trunk:
ƒ
[0] = NONE
ƒ
[1] E1 EURO ISDN
ƒ
[2] T1 CAS
ƒ
[3] T1 RAW CAS
ƒ
[4] T1 TRANSPARENT
ƒ
[5] E1 TRANSPARENT 31
ƒ
[6] E1 TRANSPARENT 30
ƒ
[7] E1 MFCR2
ƒ
[8] E1 CAS
ƒ
[9] E1 RAW CAS
ƒ
[10] T1 NI2 ISDN
ƒ
[11] T1 4ESS ISDN
ƒ
[12] T1 5ESS 9 ISDN
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ini File Field Name
Web Parameter Name
Valid Range and Description
ƒ
[13] T1 5ESS 10 ISDN
ƒ
[14] T1 DMS100 ISDN
ƒ
[15] J1 TRANSPARENT
ƒ
[16] T1 NTT ISDN = Japan - Nippon Telegraph
ƒ
[17] E1 AUSTEL ISDN = Australian Telecom
ƒ
[18] T1 HKT ISDN = Hong Kong - HKT
ƒ
[19] E1 KOR ISDN = Korean operator
ƒ
[20] T1 HKT ISDN = Hong Kong - HKT over T1
ƒ
[21] E1 QSIG
ƒ
[23] T1 QSIG
ƒ
[30] E1 FRENCH VN6 ISDN
ƒ
[31] E1 FRENCH VN3 ISDN
ƒ
[35] T1 DMS100 Meridian ISDN
ƒ
[40] E1 NI2 ISDN
ƒ
[41] E1 CAS R15
Note: The device simultaneously supports different variants of CAS
and PRI protocols on different E1/T1 spans (no more than four
simultaneous PRI variants). The device simultaneously supports
different BRI variants
Trunk Configuration
Clock Master
[ClockMaster]
Determines the Tx clock source of the E1/T1 line.
ƒ
[0] Recovered = Generate the clock according to the Rx of the
E1/T1 line (default).
ƒ
[1] Generated = Generate the clock according to the internal
TDM bus.
Notes:
Auto Clock Trunk Priority
[AutoClockTrunkPriority]
ƒ
The source of the internal TDM bus clock is determined by the
parameter TDMBusClockSource.
ƒ
For detailed information on configuring the device's clock
settings, refer to ''Clock Settings'' on page 393.
Defines the trunk priority for auto-clock fallback (per trunk
parameter).
ƒ
0 to 99 = priority (0 is the highest = default).
ƒ
100 = the SW never performs a fallback to that trunk (usually
used to mark untrusted source of clock).
Note: Fallback is enabled when the TDMBusPSTNAutoClockEnable
parameter is set to 1.
Line Code
[LineCode]
Line Build Out Loss
SIP User's Manual
Use to select B8ZS or AMI for T1 spans, and HDB3 or AMI for E1
spans.
ƒ
[0] B8ZS = use B8ZS line code (for T1 trunks only) default.
ƒ
[1] AMI = use AMI line code.
ƒ
[2] HDB3 = use HDB3 line code (for E1 trunks only).
Defines the line build out loss for the selected T1 trunk.
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ini File Field Name
Web Parameter Name
[LineBuildOut.Loss]
Valid Range and Description
ƒ
[0] 0 dB (default)
ƒ
[1] -7.5 dB
ƒ
[2] -15 dB
ƒ
[3] -22.5 dB
Note: This parameter is not applicable for PRI E1 trunks.
Trace Level
[TraceLevel]
Framing Method
[FramingMethod]
Defines the trace level:
ƒ
[0] No Trace (default)
ƒ
[1] Full ISDN Trace
ƒ
[2] Layer 3 ISDN Trace
ƒ
[3] Only ISDN Q.931 Messages Trace
ƒ
[4] Layer 3 ISDN No Duplication Trace
Determines the physical framing method for the trunk.
ƒ
[0] = default according to protocol type E1 or T1. E1 default is E1
CRC4 MultiFrame Format extended G.706B (as c); T1 default is
T1 Extended SuperFrame with CRC6 (as D).
ƒ
[1] = T1 SuperFrame Format (as B).
ƒ
[a] = E1 DoubleFrame Format
ƒ
[b] = E1 CRC4 MultiFrame Format
ƒ
[c] = E1 CRC4 MultiFrame Format extended G.706B
ƒ
[A] = T1 4-Frame multiframe.
ƒ
[B] = T1 12-Frame multiframe (D4).
ƒ
[C] = T1 Extended SuperFrame without CRC6
ƒ
[D] = T1 Extended SuperFrame with CRC6
ƒ
[E] = T1 72-Frame multiframe (SLC96)
ƒ
[F] = J1 Extended SuperFrame with CRC6 (Japan)
ISDN Configuration Parameters
ISDN Termination Side
[TerminationSide]
Selects the ISDN termination side. Applicable only to ISDN
protocols.
ƒ
[0] User side = ISDN User Termination Equipment (TE) side
(default)
ƒ
[1] Network side = ISDN Network Termination (NT) side
Note: Select 'User side' when the PSTN or PBX side is configured
as 'Network side' and vice versa. If you don't know the device's ISDN
termination side, choose 'User side'. If the D-channel alarm is
indicated, choose 'Network Side'.
NFAS Group Number
[NFASGroupNumber_x]
Indicates the NFAS group number (NFAS member) for the selected
trunk.
'x' identifies the Trunk ID.
ƒ
0 = Non NFAS trunk (default)
ƒ
1 to 9 = NFAS group number
Trunks that belong to the same NFAS group have the same number.
With ISDN Non-Facility Associated Signaling you can use single D-
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ini File Field Name
Web Parameter Name
Valid Range and Description
channel to control multiple PRI interfaces.
Notes:
NFAS Interface ID
[ISDNNFASInterfaceID_x]
ƒ
This parameter is applicable only to T1 ISDN protocols.
ƒ
For a detailed description on NFAS, refer to ''ISDN Non-Facility
Associated Signaling (NFAS)'' on page 398.
Defines a different Interface ID for each T1 trunk.
The valid range is 0 to 100. The default interface ID equals to the
trunk's ID.
'x' identifies the trunk ID.
Notes:
D-channel Configuration
[DChConfig_x]
ƒ
To set the NFAS interface ID, configure ISDNIBehavior_x to
include '512' feature per T1 trunk.
ƒ
For a detailed description on NFAS, refer to ''ISDN Non-Facility
Associated Signaling (NFAS)'' on page 398.
Defines primary, backup (optional), and B-channels only. The ini file
parameter x represents the Trunk ID.
ƒ
[0] PRIMARY= Primary Trunk (default) - contains a D-channel
that is used for signaling.
ƒ
[1] BACKUP = Backup Trunk - contains a backup D-channel that
is used if the primary D-channel fails.
ƒ
[2] NFAS = NFAS Trunk - contains only 24 B-channels, without a
signaling D-channel.
Note: This parameter is applicable only to T1 ISDN protocols.
Enable Receiving of Overlap
Dialing
[ISDNRxOverlap_x]
Enables Rx ISDN overlap per trunk ID.
ƒ
[0] Disable = Disabled (default).
ƒ
[1] Enable = Enabled.
Notes:
Local ISDN Ringback Tone
Source
[LocalISDNRBSource_ID]
ƒ
If enabled, the device receives ISDN called number that is sent in
the 'Overlap' mode.
ƒ
The SETUP message to IP is sent only after the number
(including the Sending Complete IE) is fully received (via SETUP
and/or subsequent INFO Q.931 messages).
ƒ
The MaxDigits parameter can be used to limit the length of the
collected number for device ISDN overlap dialing (if sending
complete is not received).
ƒ
If a digit map pattern is defined (DigitMapping), the device
collects digits until a match is found (e.g., for closed numbering
schemes) or until a timer expires (e.g., for open numbering
schemes). If a match is found (or the timer expires), the digit
collection process is terminated even if Sending Complete wasn't
received.
Determines whether Ringback tone is played to the ISDN by the
PBX / PSTN or by the device.
ƒ
[0] PBX = PBX / PSTN (default).
ƒ
[1] Gateway.
This parameter is applicable to ISDN protocols. It is used
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ini File Field Name
Web Parameter Name
Valid Range and Description
simultaneously with the parameter PlayRBTone2Trunk. The ID in the
ini file parameter depicts the trunk number, where 0 is the first trunk.
Progress Indicator to ISDN
[ProgressIndicator2ISDN_I
D]
Set PI in Rx Disconnect
Message
[PIForDisconnectMsg_ID]
ISDN Transfer Capabilities
[ISDNTransferCapability_ID
]
Progress Indicator (PI) to ISDN. The ID in the ini file parameter
depicts the trunk number, where 0 is the first trunk.
ƒ
[-1] Not Configured = The PI in ISDN messages is set according
to the parameter PlayRBTone2Tel (default).
ƒ
[0] No PI = PI is not sent to ISDN.
ƒ
[1] PI = 1; [8] PI = 8: The PI value is sent to PSTN in
Q.931/Proceeding and Alerting messages. Typically, the
PSTN/PBX cuts through the audio channel without playing local
Ringback tone, enabling the originating party to hear remote Call
Progress Tones or network announcements.
Defines the device's behavior when a Disconnect message is
received from the ISDN before a Connect message is received. The
ID in the ini file parameter depicts the trunk number, where 0 is the
first trunk.
ƒ
[-1] Not Configured = Sends a 183 SIP response according to the
received progress indicator (PI) in the ISDN Disconnect message.
If PI = 1 or 8, the device sends a 183 response, enabling the
PSTN to play a voice announcement to the IP side. If there isn't a
PI in the Disconnect message, the call is released (default).
ƒ
[0] No PI = Doesn't send a 183 response to IP. The call is
released.
ƒ
[1] PI = 1; [8] PI = 8: Sends a 183 response to IP.
Defines the IP-to-ISDN Transfer Capability of the Bearer Capability
IE in ISDN SETUP messages. The ID in the ini file parameter depicts
the trunk number.
ƒ
[-1] Not Configured
ƒ
[0] Audio 3.1 = Audio (default).
ƒ
[1] Speech = Speech.
ƒ
[2] Data = Data.
ƒ
Audio 7 = Currently not supported.
Note: If this parameter isn't configured or equals to '-1', Audio 3.1
capability is used.
ISDN Flexible Behavior Parameters
ISDN protocol is implemented in different Switches / PBXs by different vendors. Several
implementations vary a little from the specification. Therefore, to provide a flexible interface that
supports these ISDN variants, the ISDN behavior parameters are used.
To configure the different behavior bits in the Web interface, you can either enter the exact
hexadecimal bits value in the field to the right of the relevant parameter, or directly configure each bit
field by completing the following steps:
1. Click the arrow
button to the right of the relevant parameter; the relevant behavior page
appears.
2. Modify each bit field according to your requirements.
3. Click the Submit button to save your changes.
Q.931 Layer Response
Behavior
Version 5.6
Bit-field used to determine several behavior options that influence the
behaviour of the Q.931 protocol. To select the options, click the arrow
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ini File Field Name
Web Parameter Name
[ISDNIBehavior]
SIP User's Manual
Valid Range and Description
button, and then for each required option, select 1 to enable. The
default is 0 (i.e., disable).
ƒ
[1] NO STATUS ON UNKNOWN IE = Q.931 Status message isn't
sent if Q.931 received message contains an
unknown/unrecognized IE(s). By default, the Status message is
sent.
Note: Applicable only to PRI variants in which sending of Status
message is optional.
ƒ
[2] NO STATUS ON INV OP IE = Q.931 Status message isn't
sent if an optional IE with invalid content is received. By default,
the Status message is sent.
Note: Applicable only to PRI variants in which sending of Status
message is optional.
ƒ
[4] ACCEPT UNKNOWN FAC IE = Accepts
unknown/unrecognized Facility IE. Otherwise, the Q.931
message that contains the unknown Facility IE is rejected
(default).
Note: Applicable only to PRI variants where a complete ASN1
decoding is performed on Facility IE.
ƒ
[128] SEND USER CONNECT ACK = Connect ACK message is
sent in response to received Q.931 Connect. Otherwise, the
Connect ACK is not sent (default).
Note: Applicable only to Euro ISDN User side outgoing calls.
ƒ
[512] EXPLICIT INTERFACE ID = Enables to configure T1 NFAS
Interface ID (refer to the parameter ISDNNFASInterfaceID_x).
Note: Applicable to 4/5ESS, DMS, NI-2 and HKT variants.
ƒ
[2048] ALWAYS EXPLICIT = Always set the Channel
Identification IE to explicit Interface ID, even if the B-channel is on
the same trunk as the D-channel.
Note: Applicable to 4/5ESS, DMS and NI-2 variants.
ƒ
[32768] ACCEPT MU LAW =Mu-Law is also accepted in ETSI.
ƒ
[65536] EXPLICIT PRES SCREENING = The calling party
number (octet 3a) is always present even when presentation and
screening are at their default.
Note: Applicable only to ETSI, NI-2, and 5ESS.
ƒ
[131072] STATUS INCOMPATIBLE STATE = Clears the call on
receipt of Q.931 Status with incompatible state. Otherwise, no
action is taken (default).
ƒ
[262144] STATUS ERROR CAUSE = Clear call on receipt of
STATUS according to cause value.
ƒ
[524288] ACCEPT A LAW =A-Law is also accepted in 5ESS.
ƒ
[2097152] RESTART INDICATION
=acEV_PSTN_RESTART_CONFIRM is generated on receipt of a
RESTART message.
ƒ
[4194304] FORCED RESTART = On data link (re)initialization,
send RESTART if there is no call.
ƒ
[1073741824] NS QSI ENCODE INTEGER = If this bit is set,
INTEGER ASN.1 type is used in operator coding (compliant to
new ECMA standards); otherwise, OBJECT IDENTIFIER ASN.1
type is used.
Note: Only applicable only to QSIG.
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ini File Field Name
Web Parameter Name
Valid Range and Description
ƒ
[2147483648] NS 5ESS NATIONAL = Use the National mode of
AT&T 5ESS for B-channel maintenance.
Note: To configure the device to support several ISDNIBehavior
features, add the individual feature values. For example, to support
both [512] and [2048] features, set ISDNIBehavior = 2560 (i.e., 512 +
2048).
Outgoing Calls Behavior
[ISDNOutCallsBehavior]
This parameter determines several behaviour options that influence
the behaviour of the ISDN Stack outgoing calls. To select options,
click the arrow button, and then for each required option, select 1 to
enable. The default is 0 (i.e., disable).
ƒ
[2] USER SENDING COMPLETE =When this bit is set, the
device doesn't automatically generate the information element
Sending-Complete IE in the SETUP message. If this bit is not set,
the device generates it automatically in the SETUP message
only.
ƒ
[16] USE MU LAW = When set, the device sends G.711-m-Law
in outgoing voice calls. When disabled, the device sends G.711A-Law in outgoing voice calls. (Applicable only to the Korean
variant.)
ƒ
[128] DIAL WITH KEYPAD = When enabled, the device uses the
Keypad IE to store the called number digits instead of the
CALLED_NB IE. (Only applicable to the KOR variant (Korean
network). Useful for Korean switches that don't accept the
CALLED_NB IE.)
ƒ
[256] STORE CHAN ID IN SETUP =When this bit is set, the
device forces the sending of a Channel-Id IE in an outgoing
SETUP message even if it's not required by the standard (i.e.,
optional), and no Channel-Id has been specified in the
establishment request. This is useful for improving required
compatibility with switches. On PRI lines, it indicates an unused
channel ID, preferred only.
ƒ
[572] USE A LAW = When set, the device sends G.711 A-Law in
outgoing voice calls. When disabled, the device sends the default
G.711-Law in outgoing voice calls. Applicable to E10 variant.
ƒ
[1024] = Numbering plan / type for T1 IP-to-Tel calling numbers
are defined according to the manipulation tables or according to
the RPID header (default). Otherwise, the plan / type for T1 calls
are set according to the length of the calling number.
ƒ
[2048] = When this bit is set, the device accepts any IA5
character in the called_nb and calling_nb strings and sends any
IA5 character in the called_nb, and is not restricted to extended
digits only (i.e., 0-9,*,#).
ƒ
[16384] DLCI REVERSED OPTION = Behavior bit used in the
IUA interface groups to indicate that the reversed format of the
DLCI field must be used.
Note: When using the ini file to configure the device to support
several ISDNOutCallsBehavior features, add the individual feature
values. For example, to support both [2] and [16] features, set
ISDNOutCallsBehavior = 18 (i.e., 2 + 16).
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ini File Field Name
Web Parameter Name
Incoming Calls Behavior
[ISDNInCallsBehavior]
Valid Range and Description
This is the bit-field used to determine several behavior options that
influence how the ISDN Stack INCOMING calls behave. To select
the options, click the arrow button, and then for each required option,
select 1 to enable. The default is 0 (i.e., disable).
ƒ
[32] DATA CONN RS = Sends a CONNECT (answer) message
on NOT incoming Tel calls.
ƒ
[64] VOICE CONN RS = device sends a CONNECT (answer)
message on incoming Tel calls.
ƒ
[2048] CHAN ID IN FIRST RS = Sends Channel ID in the first
response to an incoming Q.931 Call Setup message. Otherwise,
the Channel ID is sent only if the device requires changing the
proposed Channel ID (default).
ƒ
[8192] CHAN ID IN CALL PROC = Sends Channel ID in a Q.931
Call Proceeding message.
ƒ
[65536] PROGR IND IN SETUP ACK = Includes Progress
Indicator (PI=8) in Setup ACK message if an empty called
number is received in an incoming SETUP message. Applicable
to overlap dialing mode. The parameter also directs the device to
play a dial tone (for TimeForDialTone), until the next called
number digits are received.
ƒ
[262144] = NI-2 second redirect number. You can select and use
(in INVITE messages) the NI-2 second redirect number if two
redirect numbers are received in Q.931 Setup for incoming Tel-toIP calls.
Note: When using the ini file to configure the device to support
several ISDNInCallsBehavior features, add the individual feature
values. For example, to support both [2048] and [65536] features,
set ISDNInCallsBehavior = 67584 (i.e., 2048 + 65536).
General Call Control Behavior
[ISDNGeneralCCBehavior]
SIP User's Manual
Bit-field used to determine several general CC behavior options. To
select the options, click the arrow button, and then for each required
option, select 1 to enable. The default is 0 (i.e., disable).
ƒ
[2] = data calls with interworking indication use 64 kbps Bchannels (physical only).
ƒ
[8] REVERSE CHAN ALLOC ALGO = Channel ID allocation
algorithm.
ƒ
[16] = The device clears down the call if it receives a NOTIFY
message specifying 'User-Suspended'. A NOTIFY (UserSuspended) message is used by some networks (e.g., in Italy or
Denmark) to indicate that the remote user has cleared the call,
especially in the case of a long distance voice call.
ƒ
[32] CHAN ID 16 ALLOWED = Applies only to ETSI E1 lines
(30B+D). Enables handling the differences between the newer
QSIG standard (ETS 300-172) and other ETSI-based standards
(ETS 300-102 and ETS 300-403) in the conversion of B-channel
ID values into timeslot values:
1) In 'regular ETSI' standards, the timeslot is identical to the Bchannel ID value, and the range for both is 1 to 15 and 17 to 31.
The D-channel is identified as channel-id #16 and carried into the
timeslot #16.
2) In newer QSIG standards, the channel-id range is 1 to 30, but
the timeslot range is still 1 to 15 and 17 to 31. The D-channel is
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not identified as channel-id #16, but is still carried into the
timeslot #16.
When this bit is set, the channel ID #16 is considered as a valid
B-channel ID, but timeslot values are converted to reflect the
range 1 to 15 and 17 to 31. This is the new QSIG mode of
operation. When this bit is not set (default), the channel_id #16 is
not allowed, as for all ETSI-like standards.
ƒ
[64] USE T1 PRI = PRI interface type is forced to T1.
ƒ
[128] USE E1 PRI = PRI interface type is forced to E1.
ƒ
[256] START WITH B CHAN OOS = B-channels start in the OutOf-Service state (OOS).
ƒ
[512] CHAN ALLOC LOWEST = CC allocates B-channels
starting from the lowest available B-channel id.
ƒ
[1024] CHAN ALLOC HIGHEST = CC allocates B-channels
starting from the highest available B-channel id.
Note: When using the ini file to configure the device to support
several ISDNGeneralCCBehavior features, add the individual feature
values. For example, to support both [16] and [32] features, set
ISDNGeneralCCBehavior = 48 (i.e., 16 + 32).
CAS Configuration (These parameters only appear if the protocol Type is
CAS Table
[CASTableIndex_x]
Defines CAS protocol for each trunk ID from a list of CAS protocols
defined by the parameter CASFileName_Y.
For example:
CASFileName_0 = 'E_M_WinkTable.dat'
CASFileName_1 = 'E_M_ImmediateTable.dat'
CASTableIndex_0 = 0
CASTableIndex_1 = 0
CASTableIndex_2 = 1
CASTableIndex_3 = 1
Trunks 0 and 1 use the E&M Winkstart CAS protocol, while trunks 2
and 3 use the E&M Immediate Start CAS protocol.
Note: For additional CAS table ini file parameters (CASFileName_0,
CASFileName_1, CASFileName_7, and CASTablesNum), refer to
''E1/T1 Configuration Parameters'' on page 303.
Dial Plan
[CasTrunkDialPlanName]
The Dial Plan name that is used on a specific trunk.
The range is up to 11 character strings.
Miscellaneous
PSTN Alert Timeout
[TrunkPSTNAlertTimeout_I
D]
Alert Timeout (ISDN T301 timer) in seconds for outgoing calls to
PSTN. This timer is used between the time that a SETUP message
is sent to the Tel side (IP-to-Tel call establishment) and a CONNECT
message is received. If ALERT is received, the timer is restarted.
In the ini file parameter, ID depicts the trunk number, where 0 is the
first trunk.
The range is 1 to 600. The default is 180.
Digital Out-Of-Service
Behavior
[DigitalOOSBehaviorFor
Trunk_ID]
Determines the method for setting digital trunks to Out-Of-Service
state per trunk.
Version 5.6
ƒ
[-1] Not Configured = Use the settings of the DigitalOOSBehavio
parameter for per device (default).
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ini File Field Name
Web Parameter Name
Valid Range and Description
ƒ
[0] Default = Uses default behavior for each trunk (see note
below).
ƒ
[1] Service = Sends ISDN In or Out of Service (only for ISDN
protocols that support Service message).
ƒ
[2] D-Channel = Takes D-Channel down or up (ISDN only).
ƒ
[3] Alarm = Sends or cleans PSTN AIS Alarm (ISDN and CAS).
ƒ
[4] Block = Blocks trunk (CAS only).
Notes:
Play Ringback Tone to Trunk
[PlayRBTone2Trunk_ID]
SIP User's Manual
ƒ
The default behavior (value 0) is as follows:
- ISDN: Use Service messages on supporting variants and use
Alarm on non-supporting variants.
- CAS: Use Alarm.
ƒ
When updating this parameter value at run-time, you must stop
the trunk and then restart it for the update to take effect.
ƒ
To determine the method for setting Out-Of-Service state for all
trunks (i.e., per device), use the DigitalOOSBehavior parameter
(refer to ''Configuring the Digital Gateway Parameters'' on page
207).
ƒ
The ID in the ini file parameter name represents the trunk
number, where 0 is the first trunk.
Determines the method for playing a ringback tone (RBT) to the
Trunk side. In the ini file parameter, ID depicts the Trunk number,
where 0 is the first trunk.
ƒ
[-1] = Not configured - use the value of the parameter
PlayRBTone2Tel.
ƒ
[0] Don't Play = The device configured with ISDN / CAS protocol
type, doesn't play an RBT. No PI is sent to the ISDN unless the
parameter ProgressIndicator2ISDN_ID is configured differently.
ƒ
[1] Play on Local = The device configured with CAS protocol
type, plays a local RBT to PSTN upon receipt of a 180 Ringing
response (with or without SDP). Note: Receipt of a 183 response
doesn't cause the device configured with CAS to play an RBT
(unless SIP183Behaviour = 1).
The device configured with ISDN protocol type operates
according to the parameter LocalISDNRBSource:
1) If the device receives a 180 Ringing response (with or without
SDP) and LocalISDNRBSource = 1, it plays an RBT and sends
an Alert with PI = 8 (unless the parameter
ProgressIndicator2ISDN_ID is configured differently).
2) If LocalISDNRBSource = 0, the device doesn't play an RBT
and an Alert message (without PI) is sent to the ISDN. In this
case, the PBX / PSTN should play the RBT to the originating
terminal by itself.
Note: Receipt of a 183 response doesn't cause the device with
ISDN protocol type to play an RBT; the device issues a Progress
message (unless SIP183Behaviour = 1). If SIP183Behaviour = 1,
the 183 response is treated the same way as a 180 Ringing
response.
ƒ
[2] Prefer IP = Play according to 'Early Media' (default). If a 180
response is received and the voice channel is already open (due
to a previous 183 early media response or due to an SDP in the
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current 180 response), the device with ISDN / CAS protocol type
doesn't play the RBT; PI = 8 is sent in an ISDN Alert message
(unless the parameter ProgressIndicator2ISDN_ID is configured
differently).
If a 180 response is received, but the 'early media' voice channel
is not opened, the device with CAS protocol type plays an RBT to
the PSTN. The device with ISDN protocol type operates
according to the parameter LocalISDNRBSource:
1) If LocalISDNRBSource = 1, the device plays an RBT and
sends an ISDN Alert with PI = 8 to the ISDN (unless the
parameter ProgressIndicator2ISDN_ID is configured differently).
2) If LocalISDNRBSource = 0, the device doesn't play an RBT.
No PI is sent in the ISDN Alert message (unless the parameter
ProgressIndicator2ISDN_ID is configured differently). In this case,
the PBX / PSTN should play an RBT tone to the originating
terminal by itself.
Note: Receipt of a 183 response results in an ISDN Progress
message (unless SIP183Behaviour = 1). If SIP183Behaviour = 1
(183 is handled the same way as a 180 + SDP), the device sends
an Alert message with PI = 8, without playing an RBT.
B-Channel Negotiation
[BChannelNegotiationForTr
unk_ID]
Determines the ISDN B-Channel negotiation mode.
ƒ
[-1] Not Configured = use per device configuration of
BChannelNegotiation parameter (default).
ƒ
[0] Preferred = Preferred.
ƒ
[1] Exclusive = Exclusive.
ƒ
[2] Any = Any.
Notes:
RTP Only Mode
[RTPOnlyModeForTrunk_ID
]
ƒ
Applicable to ISDN protocols.
ƒ
The option ‘Any’ is only applicable if TerminationSide is set to 0
(i.e., User side).
ƒ
The ID in the ini file parameter name represents the trunk
number, where 0 is the first trunk.
Enables the device to start sending and/or receiving RTP packets to
and from remote endpoints without the need to establish a Control
session. The remote IP address is determined according to the 'Tel
to IP Routing' table (or 'Outbound IP Routing' table if EnableSBC is
set to 1). The port is the same port as the local RTP port (configured
by the parameter BaseUDPPort and the channel on which the call is
received).
ƒ
[-1] Not Configured = Use the per device parameter
(RTPOnlyMode) value (default).
ƒ
[0] Disable = Disabled.
ƒ
[1] Transmit & Receive = send and receive RTP packets.
ƒ
[2] Transmit Only = send RTP packets only.
ƒ
[3] Receive Only = receive RTP packets only.
Note: The ID in the ini file parameter depicts the trunk number,
where 0 is the first trunk.
Version 5.6
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Web Parameter Name
Digital Out-Of-Service
Behavior
[DigitalOOSBehavior]
Valid Range and Description
Determines the method for setting digital trunks to Out-Of-Service
state per device.
ƒ
[0] Default = Uses default behavior for each trunk - see note
below (default)
ƒ
[1] Service = Sends ISDN In or Out of Service (only for ISDN
protocols that support Service message).
ƒ
[2] D-Channel = Takes D-Channel down or up (ISDN only).
ƒ
[3] Alarm = Sends or clears PSTN AIS Alarm (ISDN and CAS).
ƒ
[4] Block = Blocks trunk (CAS only).
Notes:
Transfer Mode
[TrunkTransferMode]
ƒ
The default behavior (value 0) is as follows:
- ISDN: Use Service messages on supporting variants and use
Alarm on non-supporting variants.
- CAS: Use Alarm.
ƒ
When updating this parameter value at run-time, you must stop
the trunk and then restart it for the update to take effect.
ƒ
To determine the method for setting Out-Of-Service state per
trunk, use the DigitalOOSBehaviorFor Trunk_ID parameter (refer
to ''Trunk Settings'' on page 82).
Enables the trunk Transfer Mode. Refer to TrunkTransferMode (0, 1,
or 3) in ''ISDN and CAS Interworking-Related Parameters'' on page
307.
Note: This parameter is only available for Protocol Type T1 CAS.
Enable TBCT
[TrunkTransferMode]
Enables the Two B Channel Transfer (TBCT) trunk transfer mode.
Refer to TrunkTransferMode (0 and 2) in ''ISDN and CAS
Interworking-Related Parameters'' on page 307.
Note: This parameter is only available for Protocol Type T1 N12
ISDN.
Enable RLT
[TrunkTransferMode]
Enables the Release Link Trunk (RLT) trunk transfer mode. Refer to
TrunkTransferMode (0 and 2) in ''ISDN and CAS InterworkingRelated Parameters'' on page 307.
Note: This parameter is only available for Protocol Type T1 DMS100
ISDN.
Enable Single Step Transfer
[TrunkTransferMode]
Enables the Single Step Transfer Trunk transfer mode. Refer to
TrunkTransferMode (0 and 4) in ''ISDN and CAS InterworkingRelated Parameters'' on page 307.
Enable ECT
[TrunkTransferMode]
Enables the Explicit Call Transfer (ECT) trunk transfer mode. Refer
to TrunkTransferMode (0 and 2) in ''ISDN and CAS InterworkingRelated Parameters'' on page 307.
Note: This parameter is only available for Protocol Type E1 EURO
ISDN.
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3. Web-Based Management
Configuring the CAS State Machines
The 'CAS State Machine' page allows you to modify various timers and other basic
parameters to define the initialization of the CAS state machine without changing the state
machine itself (no compilation is required). The change doesn't affect the state machine
itself, but rather the configuration.
¾ To modify the CAS state machine parameters, take these 6 steps:
1.
Open the ‘CAS State Machine' page (Configuration tab > PSTN Settings menu >
CAS State Machines page item).
Figure 3-49: CAS State Machine Page
2.
Ensure that the trunk is inactive. The trunk number displayed in the 'Related Trunks'
field must be green. If it is red (indicating that the trunk is active), click the trunk
number to open the 'Trunk Settings' page (refer to ''Configuring the Trunk Settings'' on
page 82), select the required Trunk number icon, and then click Stop Trunk.
3.
In the 'CAS State Machine' page, modify the required parameters according to the
table below.
4.
Once you have completed the configuration, activate the trunk if required in the 'Trunk
Settings' page, by clicking the trunk number in the 'Related Trunks' field, and in the
'Trunk Settings' page, select the required Trunk number icon, and then click Apply
Trunk Settings.
5.
Click Submit.
6.
Reset the device (refer to ''Resetting the Device'' on page 228).
Notes:
Version 5.6
•
It's strongly recommended that you don't modify the default values unless
you fully understand the implications of the changes and know the default
values. Every change affects the configuration of the state machine
parameters and the call process related to the trunk you are using with
this state machine.
•
You can modify CAS state machine parameters only if the following
conditions are met:
1) Trunks are inactive (stopped), i.e., the 'Related Trunks' field displays
the trunk number in green.
2) State machine is not in use or is in reset, or when it is not related to
any trunk. If it is related to a trunk, you must delete the trunk or deactivate (Stop) the trunk.
•
Field values displaying '-1' indicate CAS default values. In other words,
CAS state machine values are used.
•
The modification of the CAS state machine occurs at the CAS application
initialization only for non-default values (-1).
•
For a detailed description of the CAS Protocol table, refer to the Product
Reference Manual.
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Table 3-21: CAS State Machine Parameters Description
Parameter
Description
Generate Digit On Time
[CasStateMachineGenerateDigitOnTime]
Generates digit on-time (in msec).
The value must be a positive value. The
default value is -1.
Generate Inter Digit Time
[CasStateMachineGenerateInterDigitTime]
Generates digit off-time (in msec).
The value must be a positive value. The
default value is -1.
DTMF Max Detection Time
[CasStateMachineDTMFMaxOnDetectionTime]
Detects digit maximum on time (according
to DSP detection information event) in msec
units.
The value must be a positive value. The
default value is -1.
DTMF Min Detection Time
[CasStateMachineDTMFMinOnDetectionTime]
Detects digit minimum on time (according to
DSP detection information event) in msec
units. The digit time length must be longer
than this value to receive a detection. Any
number may be used, but the value must be
less than
CasStateMachineDTMFMaxOnDetectionTi
me.
The value must be a positive value. The
default value is -1.
MAX Incoming Address Digits
[CasStateMachineMaxNumOfIncomingAddressDigi
ts]
Defines the limitation for the maximum
address digits that need to be collected.
After reaching this number of digits, the
collection of address digits is stopped.
The value must be an integer. The default
value is -1.
MAX Incoming ANI Digits
[CasStateMachineMaxNumOfIncomingANIDigits]
Defines the limitation for the maximum ANI
digits that need to be collected. After
reaching this number of digits, the collection
of ANI digits is stopped.
The value must be an integer. The default
value is -1.
Collet ANI
[CasStateMachineCollectANI]
In some cases, when the state machine
handles the ANI collection (not related to
MFCR2), you can control the state machine
to collect ANI or discard ANI.
Digit Signaling System
[CasStateMachineDigitSignalingSystem]
SIP User's Manual
ƒ
[0] No = Don't collect ANI.
ƒ
[1] Yes = Collect ANI.
ƒ
[-1] Default = Default value.
Defines which Signaling System to use in
both directions (detection\generation).
98
ƒ
[0] DTMF = Uses DTMF signaling.
ƒ
[1] MF = Uses MF signaling (default).
ƒ
[-1] Default = Default value.
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3.4.4
3. Web-Based Management
SS7 Configuration
The SS7 Configuration menu allows you to configure the Signaling System #7 (SS7)
protocol parameters. For a detailed description of SS7 configuration, refer to the Product
Reference Manual.
3.4.5
Sigtran Configuration
The Sigtran Configuration menu allows you to configure the SIGTRAN parameters. For a
detailed description of SIGTRAN configuration, refer to the Product Reference Manual.
3.4.6
Security Settings
The Security Settings menu allows you to configure various security settings. This menu
contains the following page items:
3.4.6.1
„
Web User Accounts (refer to ''Configuring the Web User Accounts'' on page 99)
„
Web & Telnet Access List (refer to ''Configuring the Web and Telnet Access List'' on
page 102)
„
Firewall Settings (refer to ''Configuring the Firewall Settings'' on page 103)
„
Certificates (refer to ''Configuring the Certificates'' on page 105)
„
General Security Settings (refer to ''Configuring the General Security Settings'' on
page 109)
„
IPSec Table (refer to ''Configuring the IPSec Table'' on page 114)
„
IKE Table (refer to ''Configuring the IKE Table'' on page 117)
Configuring the Web User Accounts
To prevent unauthorized access to the Web interface, two Web user accounts are available
(primary and secondary) with assigned user name, password, and access level. When you
login to the Web interface, you are requested to provide the user name and password of
one of these Web user accounts. If the Web session is idle (i.e., no actions are performed)
for more than five minutes, the Web session expires and you are once again requested to
login with your user name and password. Up to five Web users can simultaneously open
(log in to) a session on the device's Web interface.
Each Web user account is composed of three attributes:
„
User name and password: enables access (login) to the Web interface.
„
Access level: determines the extent of the access (i.e., availability of pages and read /
write privileges). The available access levels and their corresponding privileges are
listed in the table below:
Version 5.6
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Table 3-22: Web User Accounts Access Levels and Privileges
Access Level
Numeric
Representation*
Security
Administrator
200
Read / write privileges for all pages.
Administrator
100
read / write privileges for all pages except
security-related pages, which are read-only.
User Monitor
50
No access to security-related and file-loading
pages; read-only access to the other pages.
This read-only access level is typically applied
to the secondary Web user account.
No Access
0
No access to any page.
Privileges
* The numeric representation of the access level is used only to define accounts in a RADIUS server
(the access level ranges from 1 to 255).
The default attributes for the two Web user accounts are shown in the following table:
Table 3-23: Default Attributes for the Web User Accounts
Account / Attribute
User Name
(Case-Sensitive)
Password
(Case-Sensitive)
Access Level
Primary Account
Admin
Admin
Security Administrator
Note: The Access Level cannot
be changed for this account
type.
Secondary Account
User
User
User Monitor
¾ To change the Web user accounts attributes, take these 4 steps:
1.
Open the 'Web User Accounts' page (Configuration tab > Security Settings menu >
Web User Accounts page item).
Figure 3-50: Web User Accounts Page (for Users with 'Security Administrator' Privileges)
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Note: If you are logged into the Web interface as the Security Administrator, both Web
user accounts are displayed on the 'Web User Accounts' page (as shown above). If
you are logged in with the secondary user account, only the details of the secondary
account are displayed on the page.
2.
To change the access level of the secondary account:
a.
From the 'Access Level' drop-down list, select the new access level.
b.
Click Change Access Level; the new access level is applied immediately.
Notes:
3.
4.
•
The access level of the primary Web user account is 'Security
Administrator', which cannot be modified.
•
The access level of the secondary account can only be modified by the
primary account user or a secondary account user with 'Security
Administrator' access level.
To change the user name of an account, perform the following:
a.
In the field 'User Name', enter the new user name (maximum of 19 case-sensitive
characters).
b.
Click Change User Name; if you are currently logged into the Web interface with
this account, the 'Enter Network Password' dialog box appears, requesting you to
enter the new user name.
To change the password of an account, perform the following:
a.
In the field 'Current Password', enter the current password.
b.
In the fields 'New Password' and 'Confirm New Password', enter the new
password (maximum of 19 case-sensitive characters).
c.
Click Change Password; if you are currently logged into the Web interface with
this account, the 'Enter Network Password' dialog box appears, requesting you to
enter the new password.
Notes:
•
•
•
•
•
•
•
•
Version 5.6
For security, it's recommended that you change the default user name
and password.
A Web user with access level 'Security Administrator' can change all
attributes of all the Web user accounts. Web users with an access level
other than 'Security Administrator' can only change their own password
and user name.
To reset the two Web user accounts' user names and passwords to
default, set the ini file parameter ResetWebPassword to 1.
To access the Web interface with a different account, click the Log off
button located on the toolbar, click any button or page item, and then reaccess the Web interface with a different user name and password.
You can set the entire Web interface to read-only (regardless of Web
user account's access level), by using the ini file parameter
DisableWebConfig (refer to ''Web and Telnet Parameters'' on page 273).
Access to the Web interface can be disabled, by setting the ini file
parameter DisableWebTask to 1. By default, access is enabled.
You can define additional Web user accounts using a RADIUS server
(refer to the Product Reference Manual).
For secured HTTP connection (HTTPS) (refer to the Product Reference
Manual).
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3.4.6.2
Configuring the Web and Telnet Access List
The 'Web & Telnet Access List' page is used to define up to ten IP addresses that are
permitted to access the device's Web and Telnet interfaces. Access from an undefined IP
address is denied. If no IP addresses are defined, this security feature is inactive and the
device can be accessed from any IP address.
The Web and Telnet Access List can also be defined using the ini file parameter
WebAccessList_x (refer to ''Web and Telnet Parameters'' on page 273).
¾ To add authorized IP addresses for Web and Telnet interfaces
access, take these 4 steps:
1.
Open the 'Web & Telnet Access List' page (Configuration tab > Security Settings
menu > Web & Telnet Access List page item).
Figure 3-51: Web & Telnet Access List Page - Add New Entry
2.
To add an authorized IP address, in the 'Add a New Authorized IP Address' field, enter
the required IP address, and then click Add New Address; the IP address you
entered is added as a new entry to the 'Web & Telnet Access List' table.
Figure 3-52: Web & Telnet Access List Table
3.
To delete authorized IP addresses, select the Delete Row check boxes corresponding
to the IP addresses that you want to delete, and then click Delete Selected
Addresses; the IP addresses are removed from the table and these IP addresses can
no longer access the Web and Telnet interfaces.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
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Notes:
3.4.6.3
•
The first authorized IP address in the list must be your PC's (terminal) IP
address; otherwise, access from your PC is denied.
•
Only delete your PC's IP address last from the 'Web & Telnet Access List'
page. If it's deleted before the last, access from your PC is denied after
it's deleted.
Configuring the Firewall Settings
The device provides an internal firewall, allowing you (the security administrator) to define
network traffic filtering rules. You can add up to 50 ordered firewall rules. For each packet
received on the network interface, the table is scanned from the top down until a matching
rule is found. This rule can either deny (block) or permit (allow) the packet. Once a rule in
the table is located, subsequent rules further down the table are ignored. If the end of the
table is reached without a match, the packet is accepted. For detailed information on the
internal firewall, refer to the Product Reference Manual.
Note: You can also configure the firewall settings using the ini file table parameter
AccessList (refer to ''Security Parameters'' on page 276).
¾ To add firewall rules, take these 5 steps:
1.
Open the 'Firewall Settings' page (Configuration tab > Security Settings menu >
Firewall Settings page item).
Figure 3-53: Firewall Settings Page
2.
In the 'Add' field, enter the index of the access rule that you want to add, and then click
Add; a new firewall rule index appears in the table.
3.
Configure the firewall rule's parameters according to the table below.
4.
Click one of the following buttons:
5.
Version 5.6
•
Apply: saves the new rule (without activating it).
•
Duplicate Rule: adds a new rule by copying a selected rule.
•
Activate: saves the new rule and activates it.
•
Delete: deletes the selected rule.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
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¾ To edit a rule, take these 4 steps:
1.
In the 'Edit Rule' column, select the rule that you want to edit.
2.
Modify the fields as desired.
3.
Click the Apply button to save the changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
¾ To activate a de-activated rule, take these 2 steps:
1.
In the 'Edit Rule' column, select the de-activated rule that you want to activate.
2.
Click the Activate button; the rule is activated.
¾ To de-activate an activated rule, take these 2 steps:
1.
In the 'Edit Rule' column, select the activated rule that you want to de-activate..
2.
Click the DeActivate button; the rule is de-activated.
¾ To delete a rule, take these 3 steps:
1.
Select the radio button of the entry you want to activate.
2.
Click the Delete Rule button; the rule is deleted.
3.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-24: Internal Firewall Parameters
Parameter
Is Rule Active
Source IP
[AccessList_Source_IP]
Description
A read-only field indicating whether the rule is active or not.
Note: After device reset, all rules are active.
IP address (or DNS name) of source network, or a specific host.
Subnet Mask
[AccessList_Net_Mask]
IP network mask - 255.255.255.255 for a single host or the
appropriate value for the source IP addresses. The IP address of the
sender of the incoming packet is bitwise ANDed with this mask and
then compared to the field 'Source IP'.
Local Port Range
[AccessList_Start_Port]
[AccessList_End_Port]
The destination UDP/TCP ports (on this device) to which packets are
sent.
The valid range is 0 to 65535.
Note: When the protocol type isn't TCP or UDP, the entire range must
be provided.
Protocol
[AccessList_Protocol]
The protocol type (e.g., UDP, TCP, ICMP, ESP or 'Any'), or the IANA
protocol number (in the range of 0 (Any) to 255).
Note: This field also accepts the abbreviated strings 'SIP' and 'HTTP'.
Specifying these strings implies selection of the TCP or UDP
protocols, and the appropriate port numbers as defined on the device.
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Parameter
Packet Size
[AccessList_Packet_Size]
Description
Maximum allowed packet size.
The valid range is 0 to 65535.
Note: When filtering fragmented IP packets, this field relates to the
overall (re-assembled) packet size, and not to the size of each
fragment.
Byte Rate
[AccessList_Byte_Rate]
Expected traffic rate (bytes per second).
Burst Bytes
[AccessList_Byte_Burst]
Tolerance of traffic rate limit (number of bytes).
Action Upon Match
[AccessList_Allow_Type]
Action upon match (i.e., 'Allow' or 'Block').
Match Count
[AccessList_MatchCount]
A read-only field providing the number of packets accepted / rejected
by the specific rule.
3.4.6.4
Configuring the Certificates
The 'Certificates' page is used for the following:
„
Replacing the server certificate (refer to ''Server Certificate Replacement'' on page
105)
„
Replacing the client certificates (refer to ''Client Certificates'' on page 108)
„
Regenerating Self-Signed Certificates (refer to ''Self-Signed Certificates'' on page 109)
„
Updating the private key (using HTTPSPkeyFileName, as described in the Product
Reference Manual).
3.4.6.4.1 Server Certificate Replacement
The device is supplied with a working Secure Socket Layer (SSL) configuration consisting
of a unique self-signed server certificate. If an organizational Public Key Infrastructure (PKI)
is used, you may wish to replace this certificate with one provided by your security
administrator.
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¾ To replace the device's self-signed certificate, take these 8 steps:
1.
Your network administrator should allocate a unique DNS name for the device (e.g.,
dns_name.corp.customer.com). This DNS name is used to access the device and
should therefore, be listed in the server certificate.
2.
Open the ‘Certificates Signing Request' page (Configuration tab > Security Settings
menu > Certificates page item).
Figure 3-54: Certificates Signing Request Page
3.
In the 'Subject Name' field, enter the DNS name, and then click Generate CSR. A
textual certificate signing request that contains the SSL device identifier is displayed.
4.
Copy this text and send it to your security provider. The security provider (also known
as Certification Authority or CA) signs this request and then sends you a server
certificate for the device.
5.
Save the certificate to a file (e.g., cert.txt). Ensure that the file is a plain-text file
containing the ‘BEGIN CERTIFICATE’ header, as shown in the example of a Base64Encoded X.509 Certificate below:
-----BEGIN CERTIFICATE----MIIDkzCCAnugAwIBAgIEAgAAADANBgkqhkiG9w0BAQQFADA/MQswCQYDVQQGEwJGUj
ETMBEGA1UEChMKQ2VydGlwb3N0ZTEbMBkGA1UEAxMSQ2VydGlwb3N0ZSBTZXJ2ZXVy
MB4XDTk4MDYyNDA4MDAwMFoXDTE4MDYyNDA4MDAwMFowPzELMAkGA1UEBhMCRlIxEz
ARBgNVBAoTCkNlcnRpcG9zdGUxGzAZBgNVBAMTEkNlcnRpcG9zdGUgU2VydmV1cjCC
ASEwDQYJKoZIhvcNAQEBBQADggEOADCCAQkCggEAPqd4MziR4spWldGRx8bQrhZkon
WnNm`+Yhb7+4Q67ecf1janH7GcN/SXsfx7jJpreWULf7v7Cvpr4R7qIJcmdHIntmf7
JPM5n6cDBv17uSW63er7NkVnMFHwK1QaGFLMybFkzaeGrvFm4k3lRefiXDmuOe+FhJ
gHYezYHf44LvPRPwhSrzi9+Aq3o8pWDguJuZDIUP1F1jMa+LPwvREXfFcUW+w==
-----END CERTIFICATE----6.
Set the parameter 'Secured Web Connection (HTTPS)' to 'HTTPS Only' (0) (refer to
''Configuring the General Security Settings'' on page 109) to ensure you have a
method of accessing the device in case the new certificate doesn’t work. Restore the
previous setting after testing the configuration.
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7.
In the 'Certificates Files' group, click the Browse button corresponding to 'Send Server
Certificate...', navigate to the cert.txt file, and then click Send File.
8.
When the loading of the certificate is complete, save the configuration (refer to ''Saving
Configuration'' on page 230) and restart the device; the Web interface uses the
provided certificate.
Notes:
•
The certificate replacement process can be repeated when necessary
(e.g., the new certificate expires).
•
It is possible to use the IP address of the device (e.g., 10.3.3.1) instead of
a qualified DNS name in the Subject Name. This is not recommended
since the IP address is subject to changes and may not uniquely identify
the device.
•
The server certificate can also be loaded via ini file using the parameter
HTTPSCertFileName.
¾ To apply the loaded certificate for IPsec negotiations, take these 2
steps:
1.
Open the ‘IKE Table’ page (refer to ''Configuring the IKE Table'' on page 117); the
'Loaded Certificates Files' group lists the newly uploaded certificates, as shown below:
Figure 3-55: IKE Table Listing Loaded Certificate Files
2.
Version 5.6
Click the Apply button to load the certificates; future IKE negotiations are now
performed using the new certificates.
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3.4.6.4.2 Client Certificates
By default, Web servers using SSL provide one-way authentication. The client is certain
that the information provided by the Web server is authentic. When an organizational PKI is
used, two-way authentication may be desired: both client and server should be
authenticated using X.509 certificates. This is achieved by installing a client certificate on
the managing PC, and loading the same certificate (in base64-encoded X.509 format) to
the device's Trusted Root Certificate Store. The Trusted Root Certificate file should contain
both the certificate of the authorized user and the certificate of the CA.
Since X.509 certificates have an expiration date and time, the device must be configured to
use NTP (refer to ''Simple Network Time Protocol Support'' on page 383) to obtain the
current date and time. Without the correct date and time, client certificates cannot work.
¾ To enable two-way client certificates, take these 5 steps:
1.
Set the parameter 'Secured Web Connection (HTTPS)' to 'HTTPS Only' (0) in
''Configuring the General Security Settings'' on page 109 to ensure you have a method
of accessing the device in case the client certificate doesn’t work. Restore the previous
setting after testing the configuration.
2.
Open the ‘Certificates Signing Request' page (refer to ''Server Certificate
Replacement'' on page 105).
3.
In the 'Certificates Files' group, click the Browse button corresponding to 'Send
"Trusted Root Certificate Store" file ...', navigate to the file, and then click Send File.
4.
When
the
operation
is
complete,
HTTPSRequireClientCertificates to 1.
5.
Save the configuration (refer to ''Saving Configuration'' on page 230), and then restart
the device.
set
the
ini
file
parameter
When a user connects to the secured Web server:
„
If the user has a client certificate from a CA that is listed in the Trusted Root Certificate
file, the connection is accepted and the user is prompted for the system password.
„
If both the CA certificate and the client certificate appear in the Trusted Root Certificate
file, the user is not prompted for a password (thus, providing a single-sign-on
experience - the authentication is performed using the X.509 digital signature).
„
If the user doesn’t have a client certificate from a listed CA, or doesn’t have a client
certificate at all, the connection is rejected.
Notes:
SIP User's Manual
•
The process of installing a client certificate on your PC is beyond the
scope of this document. For more information, refer to your Web browser
or operating system documentation, and/or consult your security
administrator.
•
The root certificate can also be loaded via ini file using the parameter
HTTPSRootFileName.
•
You can enable Online Certificate Status Protocol (OCSP) on the device
to check whether a peer's certificate has been revoked by an OCSP
server. For further information, refer to the Product Reference Manual.
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3.4.6.4.3 Self-Signed Certificates
The device is shipped with an operational, self-signed server certificate. The subject name
for this default certificate is 'ACL_nnnnnnn', where nnnnnnn denotes the serial number of
the device. However, this subject name may not be appropriate for production and can be
changed while still using self-signed certificates.
¾ To change the subject name and regenerate the self-signed
certificate, take these 4 steps:
1.
Before you begin, ensure the following:
•
You have a unique DNS name for the device (e.g.,
dns_name.corp.customer.com). This name is used to access the device and
should therefore, be listed in the server certificate.
•
No traffic is running on the device. The certificate generation process is disruptive
to traffic and should be executed during maintenance time.
2.
Open the ‘Certificates’ page (refer to ''Server Certificate Replacement'' on page 105).
3.
In the 'Subject Name' field, enter the fully-qualified DNS name (FQDN) as the
certificate subject, and then click Generate Self-signed; after a few seconds, a
message appears displaying the new subject name.
4.
Save configuration (refer to ''Saving Configuration'' on page 230), and then restart the
device for the new certificate to take effect.
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3.4.6.5
Configuring the General Security Settings
The 'General Security Settings' page is used to configure various security features.
¾ To configure the general security parameters, take these 4 steps:
1.
Open the 'General Security Settings' page (Configuration tab > Security Settings
menu > General Security Settings page item).
Figure 3-56: General Security Settings Page
2.
Configure the General Security parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
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Table 3-25: General Security Parameters
Parameter
HTTP Authentication Mode
[WebAuthMode]
Description
Determines the authentication mode for the Web interface.
ƒ
[0] Basic Mode = Basic authentication (clear text) is used
(default).
ƒ
[1] Digest When Possible = Digest authentication (MD5) is
used.
ƒ
[2] Basic if HTTPS, Digest if HTTP = Digest authentication
(MD5) is used for HTTP, and basic authentication is used
for HTTPS.
Note: When RADIUS login is enabled (i.e., the parameter
WebRADIUSLogin is set to 1), basic authentication is forced.
Secured Web Connection (HTTPS)
[HTTPSOnly]
Determines the protocol types used to access the Web
interface.
ƒ
[0] Disable = HTTP and HTTPS (default).
ƒ
[1] Enable = Unencrypted HTTP packets are blocked.
General RADIUS Settings
Enable RADIUS Access Control
[EnableRADIUS]
Use RADIUS for Web/Telnet Login
[WebRADIUSLogin]
Determines whether the RADIUS application is enabled.
ƒ
[0] Disable = RADIUS application is disabled (default).
ƒ
[1] Enable = RADIUS application is enabled.
Uses RADIUS queries for Web and Telnet interface
authentication.
ƒ
[0] Disable (default).
ƒ
[1] Enable.
When enabled, logging in to the device's Web and Telnet
embedded servers is performed via a RADIUS server. The
device contacts a predefined server and verifies the given user
name and password pair against a remote database, in a
secure manner.
Notes:
ƒ
The parameter EnableRADIUS must be set to 1.
ƒ
RADIUS authentication requires HTTP basic authentication,
meaning the user name and password are transmitted in
clear text over the network. Therefore, it's recommended to
set the parameter HttpsOnly to 1 to force the use of HTTPS,
since the transport is encrypted.
ƒ
If using RADIUS authentication when logging in to the CLI,
only the primary Web User Account (which has Security
Administration access level) can access the device's CLI
(refer to ''Configuring the Web User Accounts'' on page 99).
RADIUS Authentication Server IP
Address
[RADIUSAuthServerIP]
IP address of the RADIUS authentication server.
RADIUS Authentication Server Port
[RADIUSAuthPort]
Port number of the RADIUS authentication server.
The default value is 1645.
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Parameter
RADIUS Shared Secret
[SharedSecret]
Description
'Secret' used to authenticate the device to the RADIUS server.
Should be a cryptographically strong password.
General RADIUS Authentication
Default Access Level
[DefaultAccessLevel]
Defines the default access level for the device when the
RADIUS (authentication) response doesn't include an access
level attribute.
The valid range is 0 to 255. The default value is 200 (Security
Administrator').
Device Behavior Upon RADIUS
Timeout
[BehaviorUponRadiusTimeout]
Defines device behavior upon a RADIUS timeout.
Local RADIUS Password Cache
Mode
[RadiusLocalCacheMode]
Local RADIUS Password Cache
Timeout
[RadiusLocalCacheTimeout]
ƒ
[0] Deny Access = Denies access.
ƒ
[1] Verify Access Locally = Checks password locally
(default).
Defines the device's mode of operation regarding the timer
(configured by the parameter RadiusLocalCacheTimeout) that
determines the validity of the user name and password
(verified by the RADIUS server).
ƒ
[0] Absolute Expiry Timer = when you access a Web page,
the timeout doesn't reset but instead, continues decreasing.
ƒ
[1] Reset Timer Upon Access = upon each access to a Web
page, the timeout always resets (reverts to the initial value
configured by RadiusLocalCacheTimeout).
Defines the time (in seconds) the locally stored user name and
password (verified by the RADIUS server) are valid. When this
time expires, the user name and password become invalid and
a must be re-verified with the RADIUS server.
The valid range is 1 to 0xFFFFFF. The default value is 300 (5
minutes).
ƒ
[-1] = Never expires.
ƒ
[0] = Each request requires RADIUS authentication.
RADIUS VSA Vendor ID
[RadiusVSAVendorID]
Defines the vendor ID that the device accepts when parsing a
RADIUS response packet.
The valid range is 0 to 0xFFFFFFFF. The default value is
5003.
RADIUS VSA Access Level
Attribute
[RadiusVSAAccessAttribute]
Defines the code that indicates the access level attribute in the
Vendor Specific Attributes (VSA) section of the received
RADIUS packet.
The valid range is 0 to 255. The default value is 35.
EtherDiscover Setting
EtherDiscover Operation Mode
N/A.
IPSec Setting
Enable IP Security
[EnableIPSec]
Dead Peer Detection Mode
[IPSecDPDMode]
SIP User's Manual
Enables / disables the Internet Protocol security (IPSec) on the
device.
ƒ
[0] Disable = IPSec is disabled (default).
ƒ
[1] Enable = IPSec is enabled.
Enables the Dead Peer Detection (DPD) 'keep-alive'
mechanism (according to RFC 3706) to detect loss of peer
connectivity.
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Parameter
Description
ƒ
[0] Disabled (default).
ƒ
[1] Periodic = message exchanges at regular intervals.
ƒ
[2] On Demand = message exchanges as needed (i.e.,
before sending data to the peer). If the liveliness of the peer
is questionable, the device sends a DPD message to query
the status of the peer. If the device has no traffic to send, it
never sends a DPD message.
For detailed information on DPD, refer to the Product
Reference Manual.
TLS Settings
TLS version
[TLSVersion]
Defines the supported versions of SSL/TLS (Secure Socket
Layer/Transport Layer Security.
ƒ
[0] SSL 2.0-3.0 and TLS 1.0 = SSL 2.0, SSL 3.0, and TLS
1.0 are supported (default).
ƒ
[1] TLS 1.0 Only = only TLS 1.0 is used.
When set to 0, SSL/TLS handshakes always start with SSL 2.0
and switch to TLS 1.0 if both peers support it. When set to 1,
TLS 1.0 is the only version supported; clients attempting to
contact the device using SSL 2.0 are rejected.
TLS Client Re-Handshake Interval
[TLSReHandshakeInterval]
Defines the time interval (in minutes) between TLS ReHandshakes initiated by the device.
The interval range is 0 to 1,500 minutes. The default is 0 (i.e.,
no TLS Re-Handshake).
TLS Mutual Authentication
[SIPSRequireClientCertificate]
Determines the device's behavior when acting as a server for
TLS connections.
ƒ
[0] Disable = The device does not request the client
certificate (default).
ƒ
[1] Enable = The device requires receipt and verification of
the client certificate to establish the TLS connection.
Notes:
Peer Host Name Verification Mode
[PeerHostNameVerificationMode]
ƒ
The SIPS certificate files can be changed using the
parameters HTTPSCertFileName and
HTTPSRootFileName.
ƒ
This parameter cannot be changed on-the-fly and requires a
device reset.
Determines whether the device verifies the Subject Name of a
remote certificate when establishing TLS connections.
ƒ
[0] Disable = Disable (default).
ƒ
[1] Server Only = Verify Subject Name only when acting as
a server for the TLS connection.
ƒ
[2] Server & Client = Verify Subject Name when acting as a
server or client for the TLS connection.
When a remote certificate is received and this parameter is not
disabled, the SubjectAltName value is compared with the list of
available Proxies. If a match is found for any of the configured
Proxies, the TLS connection is established.
The comparison is performed if the SubjectAltName is either a
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Parameter
Description
DNS name (DNSName) or an IP address. If no match is found
and the SubjectAltName is marked as ‘critical’, the TLS
connection is not established. If DNSName is used, the
certificate can also use wildcards (‘*’) to replace parts of the
domain name.
If the SubjectAltName is not marked as ‘critical’ and there is no
match, the CN value of the SubjectName field is compared with
the parameter TLSRemoteSubjectName. If a match is found,
the connection is established. Otherwise, the connection is
terminated.
TLS Client Verify Server Certificate
[VerifyServerCertificate]
Determines whether the device, when acting as client for TLS
connections, verifies the Server certificate. The certificate is
verified with the Root CA information.
ƒ
[0] Disable (default).
ƒ
[1] Enable.
Note: If Subject Name verification is necessary, the parameter
PeerHostNameVerificationMode must be used as well.
TLS Remote Subject Name
[TLSRemoteSubjectName]
Defines the Subject Name that is compared with the name
defined in the remote side certificate when establishing TLS
connections.
If the SubjectAltName of the received certificate is not equal to
any of the defined Proxies Host names/IP addresses and is not
marked as 'critical', the Common Name (CN) of the Subject
field is compared with this value. If not equal, the TLS
connection is not established. If the CN uses a domain name,
the certificate can also use wildcards (‘*’) to replace parts of the
domain name.
The valid range is a string of up to 49 characters.
Note: This parameter is applicable only if the parameter
PeerHostNameVerificationMode is set to 1 or 2.
3.4.6.6
Configuring the IPSec Table
The 'IPSec Table' page allows you to configure the Security Policy Database (SPD)
parameters for IP security (IPSec).
Note: You can also configure the IPSec table using the ini file table parameter
IPSEC_SPD_TABLE (refer to ''Security Parameters'' on page 276).
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¾ To configure the IPSec SPD table, take these 5 steps:
1.
Open the ‘IPSec Table’ page (Configuration tab > Security Settings menu > IPSec
Table page item).
Figure 3-57: IPSec Table Page
2.
From the ‘Policy Index’ drop-down list, select the rule you want to edit (up to 20 policy
rules can be configured).
3.
Configure the IPSec SPD parameters according to the table below.
4.
Click the button Create; the IPSec rule is applied on-the-fly to the device.
5.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
If no IPSec methods are defined (Encryption / Authentication), the default settings, shown in
the following table are applied.
Table 3-26: Default IKE Second Phase Proposals
Proposal
Encryption
Authentication
Proposal 0
3DES
SHA1
Proposal 1
3DES
MD5
Proposal 2
DES
SHA1
Proposal 3
DES
MD5
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Table 3-27: IPSec SPD Table Configuration Parameters
Parameter Name
IPSec Mode
[IPSecMode]
Remote Tunnel IP Address
[IPSecPolicyRemoteTunnelIPAddress]
Description
Defines the IPSec mode of operation.
ƒ
[0] Transport (Default)
ƒ
[1] Tunneling
Defines the IP address of the remote
IPSec tunneling device.
Note: This parameter is available only if
the parameter IPSecMode is set to
Tunneling (1).
Remote Subnet Mask
[IPsecPolicyRemoteSubnetMask]
Defines the subnet mask of the remote
IPSec tunneling device.
The default value is 255.255.255.255
(i.e., host-to-host IPSec tunnel).
Note: This parameter is available only if
the parameter IPSecMode is set to
Tunneling (1).
Remote IP Address
[IPSecPolicyRemoteIPAddress]
Destination IP address (or FQDN) to
which the IPSec mechanism is applied.
Notes:
Local IP Address Type
[IPSecPolicyLocalIPAddressType]
ƒ
This parameter is mandatory.
ƒ
When an FQDN is used, a DNS
server must be configured
(DNSPriServerIP).
Determines the local interface to which
the encryption is applied (applicable to
multiple IPs and VLANs).
ƒ
[0] OAM = OAMP interface (default).
ƒ
[1] Control = Control interface.
Source Port
[IPSecPolicySrcPort]
Defines the source port to which the
IPSec mechanism is applied.
The default value is 0 (i.e., any port).
Destination Port
[IPSecPolicyDstPort]
Defines the destination port to which the
IPSec mechanism is applied.
The default value is 0 (i.e., any port).
Protocol
[IPSecPolicyProtocol]
Defines the protocol type to which the
IPSec mechanism is applied.
Related Key Exchange Method Index
[IPsecPolicyKeyExchangeMethodIndex]
SIP User's Manual
ƒ
0 = Any protocol (default).
ƒ
17 = UDP.
ƒ
6 = TCP.
ƒ
Any other protocol type defined by
IANA (Internet Assigned Numbers
Authority).
IPSec is
applied to
outgoing
packets
that match
the values
defined for
these
parameters.
Determines the index for the corresponding IKE entry.
Note that several policies can be associated with a
single IKE entry.
The valid range is 0 to 19. The default value is 0.
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Parameter Name
Description
IKE Second Phase Parameters (Quick Mode)
SA Lifetime (sec)
[PsecPolicyLifeInSec]
Determines the time (in seconds) that the SA negotiated
in the second IKE session (quick mode) is valid. After
the time expires, the SA is re-negotiated.
The default value is 28,800 (i.e., 8 hours).
SA Lifetime (KB)
[IPSecPolicyLifeInKB]
Determines the lifetime (in kilobytes) that the SA
negotiated in the second IKE session (quick mode) is
valid. After this size is reached, the SA is re-negotiated.
The default value is 0 (i.e., this parameter is ignored).
These lifetime parameters [SA Lifetime (sec) and SA Lifetime (KB)] determine the duration for which
an SA is valid. When the lifetime of the SA expires, it is automatically renewed by performing the IKE
second phase negotiations. To refrain from a situation where the SA expires, a new SA is negotiated
while the old one is still valid. As soon as the new SA is created, it replaces the old one. This
procedure occurs whenever an SA is about to expire.
First to Fourth Proposal Encryption Type
[IPSecPolicyProposalEncryption_X]
First to Fourth Proposal Authentication
Type
[IPSecPolicyProposalAuthentication_X]
3.4.6.7
Determines the encryption type used in the quick mode
negotiation for up to four proposals. For the ini file
parameter, X depicts the proposal number (0 to 3)).
The valid encryption values are:
ƒ
[0] None = No encryption
ƒ
[1] DES-CBC
ƒ
[2] Triple DES-CBC
ƒ
[3] AES-CBC
ƒ
Not Defined (default)
Determines the authentication protocol used in the quick
mode negotiation for up to four proposals. For the ini file
parameter, X depicts the proposal number (0 to 3).
The valid authentication values are:
ƒ
[2] HMAC-SHA-1-96
ƒ
[4] HMAC-MD5-96
ƒ
Not Defined (default)
Configuring the IKE Table
The 'IKE Table' page is used to configure the Internet Key Exchange (IKE) parameters.
Note: You can also configure the IKE table using the ini file table parameter
IPSec_IKEDB_Table (refer to ''Security Parameters'' on page 276).
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¾ To configure the IKE table, take these 5 steps:
1.
Open the ‘IKE Table’ page (Configuration tab > Security Settings menu > IKE Table
page item).
Figure 3-58: IKE Table Page
2.
From the ‘Policy Index’ drop-down list, select the peer you want to edit (up to 20 peers
can be configured).
3.
Configure the IKE parameters according to the table below. Up to two IKE main mode
proposals (Encryption / Authentication / DH group combinations) can be defined. The
same proposals must be configured for all peers.
4.
Click Create; a row is created in the IKE table.
5.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
To delete a peer from the IKE table, select it from the ‘Policy Index’ drop-down list, click the
button Delete, and then click OK at the prompt.
If no IKE methods are defined (Encryption / Authentication / DH Group), the default settings
(shown in the following table) are applied.
Table 3-28: Default IKE First Phase Proposals
Proposal
Encryption
Authentication
DH Group
Proposal 0
3DES
SHA1
1024
Proposal 1
3DES
MD5
1024
Proposal 2
3DES
SHA1
786
Proposal 3
3DES
MD5
786
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The parameters described in the following table are used to configure the first phase (main
mode) of the IKE negotiation for a specific peer. A different set of parameters can be
configured for each of the 20 available peers.
Table 3-29: IKE Table Configuration Parameters
Parameter Name
Authentication Method
[IkePolicyAuthenticationMetho
d]
Description
Determines the authentication method for IKE.
ƒ
[0] Pre-shared Key (default)
ƒ
[1] RSA Signature
Notes:
Shared Key
[IKEPolicySharedKey]
ƒ
For pre-shared key authentication, peers participating in an
IKE exchange must have a prior (out-of-band) knowledge of
the common key (see IKEPolicySharedKey parameter).
ƒ
For RSA signature authentication, peers must be loaded with
a certificate signed by a common CA. For additional
information on certificates, refer to ''Server Certificate
Replacement'' on page 105.
Determines the pre-shared key (in textual format). Both peers
must register the same pre-shared key for the authentication
process to succeed.
Notes:
ƒ
The pre-shared key forms the basis of IPSec security and
should therefore, be handled cautiously (in the same way as
sensitive passwords). It is not recommended to use the same
pre-shared key for several connections.
ƒ
Since the ini file is in plain text format, loading it to the device
over a secure network connection is recommended, preferably
over a direct crossed-cable connection from a management
PC. For added confidentiality, use the encoded ini file option
(described in ''Secured Encoded ini File'' on page 255).
ƒ
After it is configured, the value of the pre-shared key cannot
be obtained via Web interface, ini file, or SNMP (refer the
Product Reference Manual).
IKE SA LifeTime (sec)
[IKEPolicyLifeInSec]
Determines the time (in seconds) the SA negotiated in the first
IKE session (main mode) is valid. After the time expires, the SA is
re-negotiated.
The default value is 28800 (i.e., 8 hours).
IKE SA LifeTime (KB)
[IKEPolicyLifeInKB]
Determines the lifetime (in kilobytes) that the SA negotiated in the
first IKE session (main mode) is valid. After this size is reached,
the SA is re-negotiated.
The default value is 0 (i.e., this parameter is ignored).
These lifetime parameters [IKE SA LifeTime (sec) and IKE SA LifeTime (KB)] determine the duration
the SA created in the main mode phase is valid. When the lifetime of the SA expires, it's automatically
renewed by performing the IKE first phase negotiations. To refrain from a situation where the SA
expires, a new SA is negotiated while the old one is still valid. As soon as the new SA is created, it
replaces the old one. This procedure occurs whenever an SA is about to expire.
First to Fourth Proposal
Encryption Type
[IKEPolicyProposalEncryption
_X]
Version 5.6
Determines the encryption type used in the main mode
negotiation for up to four proposals. For the ini file parameter, X
depicts the proposal number (0 to 3).
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Parameter Name
First to Fourth Proposal
Authentication Type
[IKEPolicyProposalAuthenticat
ion_X ]
First to Fourth Proposal DH
Group
[IKEPolicyProposalDHGroup_
X]
3.4.7
Description
ƒ
[1] DES-CBC
ƒ
[2] Triple DES-CBC
ƒ
[3] AES-CBC
ƒ
Not Defined (default)
Determines the authentication protocol used in the main mode
negotiation for up to four proposals. For the ini file parameter, X
depicts the proposal number (0 to 3).
ƒ
[2] HMAC-SHA1-96)
ƒ
[4] HMAC-MD5-96
ƒ
Not Defined (default)
Determines the length of the key created by the DH protocol for
up to four proposals. For the ini file parameter, X depicts the
proposal number (0 to 3).
ƒ
[0] DH-786-Bit
ƒ
[1] DH-1024-Bit
ƒ
Not Defined (default)
Protocol Configuration
The Protocol Configuration menu allows you to configure the device's SIP parameters
and contains the following submenus:
3.4.7.1
„
Protocol Definition (refer to ''Configuring the Protocol Definition Parameters'' on page
120)
„
SIP Advanced Parameters (refer to ''Configuring the SIP Advanced Parameters'' on
page 151)
„
Manipulation Tables (refer to ''Configuring the Number Manipulation Tables'' on page
164)
„
Routing Tables (refer to ''Configuring the Routing Tables'' on page 171)
„
Profile Definitions (refer to ''Configuring the Profile Definitions'' on page 190)
„
Trunk/IP Group (refer to ''Configuring the Trunk and IP Groups'' on page 195)
„
Digital Gateway (refer to “Configuring the Digital Gateway Parameters” on page 207)
Configuring the Protocol Definition Parameters
The Protocol Definition submenu allows you to configure the main SIP protocol
parameters. This submenu contains the following page items:
„
SIP General Parameters (refer to ''SIP General Parameters'' on page 121)
„
Proxy & Registration (refer to ''Proxy & Registration Parameters'' on page 132)
„
Proxy Sets Table (refer to ''Proxy Sets Table'' on page 141)
„
Coders (refer to ''Coders'' on page 144)
„
DTMF & Dialing (refer to ''DTMF & Dialing Parameters'' on page 147)
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3.4.7.1.1 SIP General Parameters
The 'SIP General Parameters' page is used to configure general SIP parameters.
¾ To configure the general SIP protocol parameters, take these 4
steps:
1.
Open the 'SIP General Parameters' page (Configuration tab > Protocol
Configuration menu > Protocol Definition submenu > SIP General Parameters
page item).
Figure 3-59: SIP General Parameters Page
Version 5.6
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2.
Configure the parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-30: SIP General Parameters (Protocol Definition)
Parameter
PRACK Mode
[PRACKMode]
Description
PRACK (Provisional Acknowledgment) mechanism mode for 1xx SIP
reliable responses.
ƒ
[0] Disable
ƒ
[1] Supported (default)
ƒ
[2] Required
Notes:
Channel Select Mode
[ChannelSelectMode]
ƒ
The Supported and Required headers contain the '100rel' tag.
ƒ
The device sends PRACK messages if the 180/183 response is
received with '100rel' in the Supported or Required headers.
Port (channel) allocation algorithm for IP-to-Tel calls.
ƒ
[0] By Dest Phone Number = Selects the device's channel according to
the called number. (default.)
ƒ
[1] Cyclic Ascending = Selects the next available channel in an
ascending cyclic order. Always selects the next higher channel number
in the trunk group. When the device reaches the highest channel
number in the trunk group, it selects the lowest channel number in the
trunk group and then starts ascending again.
ƒ
[2] Ascending = Selects the lowest available channel. It always starts at
the lowest channel number in the trunk group and if that channel is not
available, selects the next higher channel.
ƒ
[3] Cyclic Descending = Selects the next available channel in
descending cyclic order. Always selects the next lower channel number
in the trunk group. When the device reaches the lowest channel number
in the trunk group, it selects the highest channel number in the trunk
group and then starts descending again.
ƒ
[4] Descending = Selects the highest available channel. Always starts at
the highest channel number in the trunk group and if that channel is not
available, selects the next lower channel.
ƒ
[5] Dest Number + Cyclic Ascending = First selects the device's port
according to the called number. If the called number isn't found, it then
selects the next available channel in ascending cyclic order. Note that if
the called number is found, but the port associated with this number is
busy, the call is released.
ƒ
[6] By Source Phone Number = Selects the device's channel according
to the calling number.
ƒ
[7] Trunk Cyclic Ascending = Selects the device's port from the first
channel of the next trunk (next to the trunk from which the previous
channel was allocated.
Notes:
SIP User's Manual
ƒ
The internal numbers of the device's B-channels are defined by the
TrunkGroup parameter.
ƒ
For defining the channel select mode per Trunk Group, refer to
''Configuring the Trunk Group Settings'' on page 197.
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Parameter
Enable Early Media
[EnableEarlyMedia]
Description
Enables the device to send a 183 Session Progress response with SDP
(instead of 180 Ringing), allowing the media stream to be established prior
to the answering of the call.
ƒ
[0] Disable = Early Media is disabled (default).
ƒ
[1] Enable = Enables Early Media.
Sending a 183 response depends on the Progress Indicator (PI). It is sent
only if PI is set to 1 or 8 are received in Proceeding or Alert PRI messages.
For CAS devices, see the ProgressIndicator2IP parameter.
183 Message Behavior
[SIP183Behaviour]
Defines the ISDN message that is sent when the 183 Session Progress
message is received for IP-to-Tel calls.
ƒ
[0] Progress = The device sends a PROGRESS message (default).
ƒ
[1] Alert = The device sends an ALERT message (upon receipt of a 183
response) instead of an ISDN PROGRESS message.
Session-Expires Time
[SIPSessionExpires]
Determines the numerical value that is sent in the Session-Expires header
in the first INVITE request or response (if the call is answered).
The valid range is 1 to 86,400 sec. The default is 0 (i.e., the SessionExpires header is disabled).
Minimum SessionExpires
[MinSE]
Defines the time (in seconds) that is used in the Min-SE header. This
header defines the minimum time that the user agent refreshes the
session.
The valid range is 10 to 100,000. The default value is 90.
Session Expires
Method
[SessionExpiresMeth
od]
Determines the SIP method used for session-timer updates.
ƒ
[0] Re-INVITE = Uses Re-INVITE messages for session-timer updates
(default).
ƒ
[1] UPDATE = Uses UPDATE messages.
Notes:
Asserted Identity Mode
[AssertedIdMode]
ƒ
The device can receive session-timer refreshes using both methods.
ƒ
The UPDATE message used for session-timer is excluded from the
SDP body.
Determines whether P-Asserted-Identity or P-Preferred-Identity is used in
the generated INVITE request for Caller ID (or privacy).
ƒ
[0] Disabled = None (default)
ƒ
[1] Adding PAsserted Identity
ƒ
[2] Adding PPreferred Identity
The Asserted ID mode defines the header (P-Asserted-Identity or PPreferred-Identity) that is used in the generated INVITE request. The
header also depends on the calling Privacy (allowed or restricted).
The P-Asserted-Identity (or P-Preferred-Identity) headers are used to
present the originating party's Caller ID. The Caller ID is composed of a
Calling Number and (optionally) a Calling Name.
P-Asserted-Identity (or P-Preferred-Identity) headers are used together
with the Privacy header. If Caller ID is restricted (P-Asserted-Identity is not
sent), the Privacy header includes the value 'id' ('Privacy: id'). Otherwise,
for allowed Caller ID, 'Privacy: none' is used. If Caller ID is restricted
(received from PSTN), the From header is set to
<[email protected]>.
The logic for filling the calling party parameters is as follows: the SIP
header is selected first from which the calling party parameters are
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Parameter
Description
obtained: first priority is P-Asserted-Identity, second is Remote-Party-ID,
and third is the From header. Once a URL is selected, all the calling party
parameters are set from this header. If P-Asserted-Identity is selected, the
Privacy header is checked and if the Privacy is set to 'id', the calling
number is assumed restricted.
Fax Signaling Method
[IsFaxUsed]
Determines the SIP signaling method for establishing and transmitting a fax
session after a fax is detected.
ƒ
[0] No Fax = No fax negotiation using SIP signaling. Fax transport
method is according to the parameter FaxTransportMode (default).
ƒ
[1] T.38 Relay = Initiates T.38 fax relay.
ƒ
[2] G.711 Transport = Initiates fax / modem using the coder G.711 Alaw/μ-law with adaptations (refer to Note below).
ƒ
[3] Fax Fallback = Initiates T.38 fax relay. If the T.38 negotiation fails,
the device re-initiates a fax session using the coder G.711 A-law/μ-law
with adaptations (refer to the Note below).
Notes:
Detect Fax on Answer
Tone
[DetFaxOnAnswerTon
e]
ƒ
Fax adaptations (for options 2 and 3):
Echo Canceller = On
Silence Compression = Off
Echo Canceller Non-Linear Processor Mode = Off
Dynamic Jitter Buffer Minimum Delay = 40
Dynamic Jitter Buffer Optimization Factor = 13
ƒ
If the device initiates a fax session using G.711 (option 2 and possibly
3), a 'gpmd' attribute is added to the SDP in the following format:
For A-law: 'a=gpmd:8 vbd=yes;ecan=on'.
For μ-law: 'a=gpmd:0 vbd=yes;ecan=on'.
ƒ
When IsFaxUsed is set to 1, 2, or 3, the parameter FaxTransportMode
is ignored.
ƒ
When the value of IsFaxUsed is other than 1, T.38 might still be used
without the control protocol's involvement. To completely disable T.38,
set FaxTransportMode to a value other than 1.
ƒ
For detailed information on fax transport methods, refer to ''Fax/Modem
Transport Modes'' on page 351.
Determines when the device initiates a T.38 session for fax transmission.
ƒ
[0] Initiate T.38 on Preamble = The device to which the called fax is
connected initiates a T.38 session on receiving Preamble signal from
the fax (default).
ƒ
[1] Initiate T.38 on CED = The device to which the called fax is
connected initiates a T.38 session on receiving a CED answer tone
from the fax. This option can only be used to relay fax signals, as the
device sends T.38 Re-INVITE on detection of any fax/modem Answer
tone (2100 Hz, amplitude modulated 2100 Hz, or 2100 Hz with phase
reversals). The modem signal fails when using T.38 for fax relay.
Notes:
SIP User's Manual
ƒ
For this parameter to take effect, you must reset the device.
ƒ
This parameters is applicable only if the ini file parameter IsFaxUsed is
set to 1 or 3.
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Parameter
SIP Transport Type
[SIPTransportType]
Description
Determines the default transport layer for outgoing SIP calls initiated by the
device.
ƒ
[0] UDP (default)
ƒ
[1] TCP
ƒ
[2] TLS (SIPS)
Notes:
ƒ
It's recommended to use TLS for communication with a SIP Proxy and
not for direct device-to-device communication.
ƒ
For received calls (i.e., incoming), the device accepts all these
protocols.
ƒ
The value of this parameter is also used by the SAS application as the
default transport layer for outgoing SIP calls.
SIP UDP Local Port
[LocalSIPPort]
Local UDP port for SIP messages.
The valid range is 1 to 65534. The default value is 5060.
SIP TCP Local Port
[TCPLocalSIPPort]
Local TCP port for SIP messages.
The valid range is 1 to 65534. The default value is 5060.
SIP TLS Local Port
[TLSLocalSIPPort]
Local TLS port for SIP messages.
The valid range is 1 to 65534. The default value is 5061.
Note: The value of must be different than the value of 'SIP TCP Local Port'
(TCPLocalSIPPort).
Enable SIPS
[EnableSIPS]
Enables secured SIP (SIPS URI) connections over multiple hops.
ƒ
[0] Disable (default).
ƒ
[1] Enable.
When 'SIP Transport Type' is set to TLS (SIPTransportType = 2) and
'Enable SIPS' is disabled, TLS is used for the next network hop only. When
'SIP Transport Type' is set to TCP or TLS (SIPTransportType = 2 or 1) and
'Enable SIPS' is enabled, TLS is used through the entire connection (over
multiple hops).
Note: If this parameter is enabled and 'SIP Transport Type' is set to UDP
(SIPTransportType = 0), the connection fails.
Enable TCP
Connection Reuse
[EnableTCPConnectio
nReuse]
Enables the reuse of the same TCP connection for all calls to the same
destination.
TCP Timeout
[SIPTCPTimeout]
Defines the Timer B (INVITE transaction timeout timer) and Timer F (nonINVITE transaction timeout timer), as defined in RFC 3261, when the SIP
Transport Type is TCP.
The valid range is 0 to 40 sec. The default value is 64*SIPT1Rtx msec.
SIP Destination Port
[SIPDestinationPort]
SIP destination port for sending initial SIP requests.
The valid range is 1 to 65534. The default port is 5060.
Note: SIP responses are sent to the port specified in the Via header.
Use user=phone in SIP
URL
[IsUserPhone]
Determines whether to add 'user=phone' string in SIP URI.
Version 5.6
ƒ
[0] Disable = Use a separate TCP connection for each call (default).
ƒ
[1] Enable = Use the same TCP connection for all calls.
ƒ
[0] No = 'user=phone' string isn't used in SIP URI.
ƒ
[1] Yes = 'user=phone' string is part of the SIP URI (default).
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Parameter
Use user=phone in
From Header
[IsUserPhoneInFrom]
Description
Determines whether to add 'user=phone' string in the From header.
ƒ
[0] No = Doesn't use 'user=phone' string in From header (default).
ƒ
[1] Yes = 'user=phone' string is part of the From header.
Use Tel URI for
Asserted Identity
[UseTelURIForAssert
edID]
Determines the format of the URI in the P-Asserted-Identity and PPreferred-Identity headers.
Tel to IP No Answer
Timeout
[IPAlertTimeout]
Defines the time (in seconds) that the device waits for a 200 OK response
from the called party (IP side) after sending an INVITE message. If the
timer expires, the call is released.
The valid range is 0 to 3600. The default value is 180.
Enable Remote Party
ID
[EnableRPIheader]
Enables Remote-Party-ID (RPI) headers for calling and called numbers for
Tel-to-IP calls.
Add Number Plan and
Type to RPI Header
[AddTON2RPI]
ƒ
[0] Disable = 'sip:' (default).
ƒ
[1] Enable = 'tel:'.
ƒ
[0] Disable (default).
ƒ
[1] Enable = RPI headers are generated in SIP INVITE messages for
both called and calling numbers.
Determines whether the TON/PLAN parameters are included in the
Remote-Party-ID (RPID) header.
ƒ
[0] No
ƒ
[1] Yes (default)
If RPID header is enabled (EnableRPIHeader = 1) and AddTON2RPI = 1,
it's possible to configure the calling and called number type and number
plan using the Number Manipulation tables for Tel-to-IP calls.
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Parameter
Enable History-Info
Header
[EnableHistoryInfo]
Description
Enables usage of the History-Info header.
ƒ
[0] Disable = Disable (default)
ƒ
[1] Enable = Enable
User Agent Client (UAC) Behavior:
ƒ
Initial request: The History-Info header is equal to the Request URI. If a
PSTN Redirect number is received, it is added as an additional HistoryInfo header with an appropriate reason.
ƒ
Upon receiving the final failure response, the device copies the HistoryInfo as is, adds the reason of the failure response to the last entry, and
concatenates a new destination to it (if an additional request is sent).
The order of the reasons is as follows:
1. Q.850 Reason
2. SIP Reason
3. SIP Response code
Upon receiving the final response (success or failure), the device
searches for a Redirect reason in the History-Info (i.e., 3xx/4xx SIP
reason). If found, it is passed to ISDN according to the following table:
SIP Reason Code
ISDN Redirecting Reason
ƒ
302 - Moved Temporarily
Call Forward Universal (CFU)
408 - Request Timeout
Call Forward No Answer (CFNA)
480 - Temporarily Unavailable
487 - Request Terminated
486 - Busy Here
Call Forward Busy (CFB)
600 - Busy Everywhere
ƒ
If history reason is a Q.850 reason, it is translated to the SIP reason
(according to the SIP-ISDN tables) and then to ISDN Redirect reason
according to the table above.
User Agent Server (UAS) Behavior:
Use Source Number as
Display Name
[UseSourceNumberA
sDisplayName]
Version 5.6
ƒ
The History-Info header is sent only in the final response.
ƒ
Upon receiving a request with History-Info, the UAS checks the policy in
the request. If 'session', 'header', or 'history' policy tag is found, the
(final) response is sent without History-Info; otherwise, it is copied from
the request.
Determines the use of Tel Source Number and Display Name for Tel-to-IP
calls.
ƒ
[0] No = If a Tel Display Name is received, the Tel Source Number is
used as the IP Source Number and the Tel Display Name is used as the
IP Display Name. If no Display Name is received from the Tel side, the
IP Display Name remains empty (default).
ƒ
[1] Yes = If a Tel Display Name is received, the Tel Source Number is
used as the IP Source Number and the Tel Display Name is used as the
IP Display Name. If no Display Name is received from the Tel side, the
Tel Source Number is used as the IP Source Number and also as the IP
Display Name.
ƒ
[2] Overwrite = The Tel Source Number is used as the IP Source
Number and also as the IP Display Name (even if the received Tel
Display Name is not empty).
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Parameter
Use Display Name as
Source Number
[UseDisplayNameAsS
ourceNumber]
Description
Determines the use of Source Number and Display Name for IP-to-Tel
calls.
ƒ
[0] No = If IP Display Name is received, the IP Source Number is used
as the Tel Source Number and the IP Display Name is used as the Tel
Display Name. If no Display Name is received from IP, the Tel Display
Name remains empty (default).
ƒ
[1] Yes = If an IP Display Name is received, it is used as the Tel Source
Number and also as the Tel Display Name, and Presentation is set to
Allowed (0). If no Display Name is received from IP, the IP Source
Number is used as the Tel Source Number and Presentation is set to
Restricted (1).
For example: When 'from: 100 <sip:[email protected]>' is received,
the outgoing Source Number and Display Name are set to '100' and the
Presentation is set to Allowed (0).
When 'from: <sip:[email protected]>' is received, the outgoing Source
Number is set to '100' and the Presentation is set to Restricted (1).
Enable Contact
Restriction
[EnableContactRestri
ction]
Play Ringback Tone to
IP
[PlayRBTone2IP]
Determines whether the device sets the Contact header of outgoing
INVITE requests to ‘anonymous’ for restricted calls.
ƒ
[0] = Disabled (default)
ƒ
[1] = Enabled
Determines whether or not the device plays a ringback tone (RBT) to the IP
side of the call (IP-to-Tel calls).
ƒ
[0] Don't Play = Ringback tone isn't played (default).
ƒ
[1] Play = Ringback tone is played after SIP 183 session progress
response is sent.
If configured to 1 ('Play') and EnableEarlyMedia = 1, the device plays a
ringback tone according to the following:
ƒ
For CAS interfaces: the device opens a voice channel, sends a
183+SDP response, and then plays a ringback tone to IP.
ƒ
For ISDN interfaces: if a Progress or an Alert message with PI (1 or 8) is
received from the ISDN, the device opens a voice channel, sends a
183+SDP or 180+SDP response, but doesn't play a ringback tone to IP.
If PI (1 or 8) is received from the ISDN, the device assumes that
ringback tone is played by the ISDN switch. Otherwise, the device plays
a ringback tone to IP after receiving an Alert message from the ISDN. It
sends a 180+SDP response, signaling to the calling party to open a
voice channel to hear the played ringback tone.
Notes:
Play Ringback Tone to
Tel
[PlayRBTone2Tel]
SIP User's Manual
ƒ
To enable the device to send a 183/180+SDP responses, set
EnableEarlyMedia to 1.
ƒ
If EnableDigitDelivery = 1, the device doesn't play a ringback tone to IP
and doesn't send 183 or 180+SDP responses.
Determines the method used to play a ringback tone to the Tel side. It
applies to all trunks that are not configured by the parameter
PlayRBTone2Trunk. Similar description as the parameter
PlayRBTone2Trunk.
ƒ
[0] Don't Play = Ringback tone isn't played.
ƒ
[1] Play Local = Ringback tone is played to the Tel side of the call when
180/183 response is received.
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Parameter
Description
ƒ
Use Tgrp Information
[UseSIPTgrp]
Enable GRUU
[EnableGRUU]
[2] Play According to Early Media = Ringback tone is played to the Tel
side of the call if no SDP is received in 180/183 responses. If 180/183
with SDP message is received, the device cuts through the voice
channel and doesn't play ringback tone (default).
Determines whether the SIP 'tgrp' parameter, which specifies the Trunk
Group to which the call belongs is used, according to RFC 4904.
For example:
INVITE sip::+16305550100;tgrp=1;[email protected];user=phone SIP/2.0
ƒ
[0] Disable = The 'tgrp' parameter isn't used (default).
ƒ
[1] Send Only = The Trunk Group number is added to the 'tgrp'
parameter value in the Contact header of outgoing SIP messages. If a
Trunk Group number is not associated with the call, the 'tgrp' parameter
isn't included. If a 'tgrp' value is specified in incoming messages, it is
ignored.
ƒ
[2] Send and Receive = The functionality of outgoing SIP messages is
identical to the functionality described in option (1). In addition, for
incoming SIP messages, if the Request-URI includes a 'tgrp' parameter,
the device routes the call according to that value (if possible). If the
Contact header includes a 'tgrp' parameter, it is copied to the
corresponding outgoing messages in that dialog.
Determines whether the Globally Routable User Agent URIs (GRUU)
mechanism is used.
ƒ
[0] Disable = Disable (default)
ƒ
[1] Enable = Enable
The device obtains a GRUU by generating a normal REGISTER request.
This request contains a Supported header with the value 'gruu'. The device
includes a '+sip.instance' Contact header parameter for each contact for
which the GRUU is desired. This Contact parameter contains a globally
unique ID that identifies the device instance.
The global unique ID is as follows:
ƒ
If registration is per endpoint (AuthenticationMode=0), it is the MAC
address of the device concatenated with the phone number of the
endpoint.
ƒ
If the registration is per device (AuthenticationMode=1) it is only the
MAC address.
ƒ
When the User Information mechanism is used, the globally unique ID is
the MAC address concatenated with the phone number of the endpoint
(defined in the User-Info file).
If the Registrar/Proxy supports GRUU, the REGISTER responses contain
the 'gruu' parameter in each Contact header field. The Registrar/Proxy
provides the same GRUU for the same AOR and instance-id in case of
sending REGISTER again after expiration of the registration.
The device places the GRUU in any header field which contains a URI. It
uses the GRUU in the following messages: INVITE requests, 2xx
responses to INVITE, SUBSCRIBE requests, 2xx responses to
SUBSCRIBE, NOTIFY requests, REFER requests, and 2xx responses to
REFER.
Note: If the GRUU contains the 'opaque' URI parameter, the device
obtains the AOR for the user by stripping the parameter. The resulting URI
is the AOR.
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Parameter
Description
For example:
AOR: sip:[email protected]
GRUU: sip:[email protected];opaque="kjh29x97us97d"
User-Agent Information
[UserAgentDisplayInf
o]
Defines the string that is used in the SIP request header User-Agent and
SIP response header Server. If not configured, the default string
'AudioCodes product-name s/w-version' is used (e.g., User-Agent:
Audiocodes-Sip-Gateway-Mediant 2000/v.5.40.010.006). When configured,
the string 'UserAgentDisplayInfo s/w-version' is used (e.g., User-Agent:
MyNewOEM/v.5.40.010.006). Note that the version number can't be
modified.
The maximum string length is 50 characters.
SDP Session Owner
[SIPSDPSessionOwn
er]
Determines the value of the Owner line ('o' field) in outgoing SDP
messages.
The valid range is a string of up to 39 characters. The default value is
'AudiocodesGW'.
For example: o=AudiocodesGW 1145023829 1145023705 IN IP4
10.33.4.126
Play Busy Tone to Tel
[PlayBusyTone2ISDN]
Enables the device to play a busy or reorder tone to the PSTN after a Telto-IP call is released.
ƒ
[0] Don't Play = Immediately sends an ISDN Disconnect message
(default).
ƒ
[1] Play when Disconnecting = Sends an ISDN Disconnect message
with PI = 8 and plays a busy or reorder tone to the PSTN (depending on
the release cause).
ƒ
[2] Play before Disconnect = Delays the sending of an ISDN Disconnect
message for a user-defined time (configured by the
TimeForReorderTone parameter) and plays a busy or reorder tone to
the PSTN. Applicable only if the call is released from the IP [Busy Here
(486) or Not Found (404)] before it reaches the Connect state;
otherwise, the Disconnect message is sent immediately and no tones
are played.
Subject
[SIPSubject]
Defines the value of the Subject header in outgoing INVITE messages. If
not specified, the Subject header isn't included (default).
The maximum length is up to 50 characters.
Multiple Packetization
Time Format
[MultiPtimeFormat]
Determines whether the 'mptime' attribute is included in the outgoing SDP.
ƒ
[0] None = Disabled (default)
ƒ
[1] PacketCable = includes the 'mptime' attribute in the outgoing SDP -PacketCable-defined format
The 'mptime' attribute enables the device to define a separate
Packetization period for each negotiated coder in the SDP. The 'mptime'
attribute is only included if this parameter is enabled, even if the remote
side includes it in the SDP offer. Upon receipt, each coder receives its
'ptime' value in the following precedence: from 'mptime' attribute, from
'ptime' attribute, and then from default value.
Enable Semi-Attended
Transfer
[EnableSemiAttended
Transfer]
SIP User's Manual
Determines the device behavior when Transfer is initiated while in Alerting
state.
ƒ
[0] Disable = Send REFER with Replaces (default).
ƒ
[1] Enable = Send CANCEL, and after a 487 response is received, send
REFER without Replaces.
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Parameter
3xx Behavior
[3xxBehavior]
Enable P-Charging
Vector
[EnablePChargingVec
tor]
Enable VoiceMail URI
[EnableVMURI]
Description
Determines the device's behavior regarding call identifiers when a 3xx
response is received for an outgoing INVITE request. The device can either
use the same call identifiers (Call-ID, Branch, To, and From tags) or
change them in the new initiated INVITE.
ƒ
[0] Forward = Use different call identifiers for a redirected INVITE
message (default).
ƒ
[1] Redirect = Use the same call identifiers.
Enables the addition of a P-Charging-Vector header to all outgoing INVITE
messages.
ƒ
[0] Disable = Disable (default)
ƒ
[1] Enable = Enable
Enables or disables the interworking of target and cause for redirection
from Tel to IP and vice versa, according to RFC 4468.
ƒ
[0] Disable = Disable (default).
ƒ
[1] Enable = Enable
Upon receipt of an ISDN SETUP message with redirect values, the device
maps the Redirect phone number to the SIP 'target' parameter and the
Redirect number reason to the SIP 'cause' parameter in the Request-URI.
Redirecting Reason
>> SIP Response Code
Unknown
>> 404
User busy
>> 486
No reply
>> 408
Deflection
>> 487/480
Unconditional
>> 302
Others
>> 302
If the device receives a Request-URI that includes a 'target' and 'cause'
parameter, the 'target' is mapped to the Redirect phone number and the
'cause' is mapped to Redirect number reason.
Retry-After Time
[RetryAfterTime]
Determines the time (in seconds) used in the Retry-After header when a
503 (Service Unavailable) response is generated by the device.
The time range is 0 to 3,600. The default value is 0.
Enable P-AssociatedURI Header
[EnablePAssociatedU
RIHeader]
Determines the device usage of the P-Associated-URI header. This header
can be received in 200 OK responses to REGISTER requests. When
enabled, the first URI in the P-Associated-URI header is used in
subsequent requests as the From / P-Asserted-Id headers value.
ƒ
[0] Disable (default).
ƒ
[1] Enable.
Note: P-Associated-URIs in registration responses is handled only if the
device is registered per endpoint (using the User Information file).
Source Number
Preference
[SourceNumberPrefer
ence]
Version 5.6
Determines the SIP header used to determine the Source Number in
incoming INVITE messages.
ƒ
“” (empty string) = Use device's internal logic for header preference
(default).
ƒ
“FROM” = Use the Source Number received in the From header.
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Parameter
Description
The valid range is a string of up to 10 characters. The default is an empty
string.
Forking Handling Mode
[ForkingHandlingMod
e]
Determines how the device reacts to forking of outgoing INVITE messages
by the Proxy.
ƒ
[0] Sequential handling = The device opens a voice stream toward the
first 18x SIP response that includes an SDP, and disregards any 18x
response with an SDP received thereafter (default).
ƒ
[1] Parallel handling = The device opens a voice stream toward the first
18x SIP response that includes an SDP, and re-opens the stream
toward any subsequent 18x responses with an SDP.
Note: Regardless of the ForkingHandlingMode value, once a SIP 200 OK
response is received, the device uses the RTP information and re-opens
the voice stream, if necessary.
Enable Reason Header
[EnableReasonHeade
r]
Enables / disables the usage of the SIP Reason header.
ƒ
[0] Disable.
ƒ
[1] Enable (default).
Retransmission Parameters
SIP T1 Retransmission
Timer [msec]
[SipT1Rtx]
The time interval (in msec) between the first transmission of a SIP
message and the first retransmission of the same message.
The default is 500.
Note: The time interval between subsequent retransmissions of the same
SIP message starts with SipT1Rtx and is multiplied by two until SipT2Rtx.
For example (assuming that SipT1Rtx = 500 and SipT2Rtx = 4000):
ƒ
The first retransmission is sent after 500 msec.
ƒ
The second retransmission is sent after 1000 (2*500) msec.
ƒ
The third retransmission is sent after 2000 (2*1000) msec.
ƒ
The fourth retransmission and subsequent retransmissions until
SIPMaxRtx are sent after 4000 (2*2000) msec.
SIP T2 Retransmission
Timer [msec]
[SipT2Rtx]
The maximum interval (in msec) between retransmissions of SIP
messages.
The default is 4000.
Note: The time interval between subsequent retransmissions of the same
SIP message starts with SipT1Rtx and is multiplied by two until SipT2Rtx.
SIP Maximum RTX
[SIPMaxRtx]
Maximum number of UDP transmissions (first transmission plus
retransmissions) of SIP messages.
The range is 1 to 30. The default value is 7.
3.4.7.1.2 Proxy & Registration Parameters
The 'Proxy & Registration' page allows you to configure parameters that are associated with
Proxy and Registration.
Note: To view whether the device or its endpoints have registered to a SIP
Registrar/Proxy server, refer to Registration Status.
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¾ To configure the Proxy & Registration parameters, take these 4
steps:
1.
Open the 'Proxy & Registration' page (Configuration tab > Protocol Configuration
menu > Protocol Definition submenu > Proxy & Registration page item).
Figure 3-60: Proxy & Registration Page
2.
Configure the Proxy and Registration parameters according to the following table.
3.
Click the Submit button to save your changes, or click the Register or Un-Register
buttons to save your changes and register / unregister to a Proxy / Registrar.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Version 5.6
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Table 3-31: Proxy & Registration Parameters
Parameter
Description
Proxy Parameters
Use Default Proxy
[IsProxyUsed]
Enables the use of a SIP Proxy server.
ƒ
[0] No = Proxy isn't used - the internal routing table is used
instead (default).
ƒ
[1] Yes = Proxy is used. Parameters relevant to Proxy
configuration are displayed.
If you are using a Proxy server, enter the IP address of the Proxy
server in the 'Proxy Sets table' (refer to ''Proxy Sets Table'' on page
141). If you are not using a Proxy server, you must configure the
device's 'Tel to IP Routing' table (described in ''Tel to IP Routing
Table'' on page 175) or 'Outbound IP Routing' table if EnableSBC
is set to 1 (refer to “Outbound IP Routing Table” on page 178).
Proxy Set Table (button)
button to open the 'Proxy Sets
Click the right-pointing arrow
Table' page to configure groups of proxy addresses. Alternatively,
you can open this page from the Proxy Sets Table page item
(refer to ''Proxy Sets Table'' on page 141 for a description of this
page).
Note: This button appears only if the 'Use Default Proxy' parameter
is enabled.
Proxy Name
[ProxyName]
Defines the Home Proxy Domain Name. If specified, the Proxy
Name is used as the Request-URI in REGISTER, INVITE, and
other SIP messages, and as the host part of the To header in
INVITE messages. If not specified, the Proxy IP address is used
instead.
The value must be string of up to 49 characters.
Redundancy Mode
[ProxyRedundancyMode]
Determines whether the device switches back to the primary Proxy
after using a redundant Proxy.
ƒ
[0] Parking = device continues working with a redundant (now
active) Proxy until the next failure, after which it works with the
next redundant Proxy (default).
ƒ
[1] Homing = device always tries to work with the primary Proxy
server (i.e., switches back to the primary Proxy whenever it's
available).
Note: To use ProxyRedundancyMode, enable Keep-alive with
Proxy option (EnableProxyKeepAlive = 1 or 2).
Proxy IP List Refresh Time
[ProxyIPListRefreshTime]
Defines the time interval (in seconds) between each Proxy IP list
refresh.
The range is 5 to 2,000,000. The default interval is 60.
Enable Fallback to Routing
Table
[IsFallbackUsed]
Determines whether the device falls back to the 'Tel to IP Routing'
table (or 'Outbound IP Routing' table if EnableSBC is set to 1) for
call routing when Proxy servers are unavailable.
ƒ
[0] Disable = Fallback is not used (default).
ƒ
[1] Enable = 'Tel to IP Routing' table (or 'Outbound IP Routing'
table) is used when Proxy servers are unavailable.
When the device falls back to its 'Tel to IP Routing' table (or
'Outbound IP Routing' table), the device continues scanning for a
Proxy. When the device locates an active Proxy, it switches from
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Parameter
Description
internal routing back to Proxy routing.
Note: To enable the redundant Proxies mechanism, set the
parameter EnableProxyKeepAlive to 1 or 2.
Prefer Routing Table
[PreferRouteTable]
Use Routing Table for Host
Names and Profiles
[AlwaysUseRouteTable]
Determines if the device's internal routing table takes precedence
over a Proxy for routing calls.
ƒ
[0] No = Only a Proxy server is used to route calls (default).
ƒ
[1] Yes = The device checks the routing rules in the 'Tel to IP
Routing' table (or 'Outbound IP Routing' table if EnableSBC is
set to 1) for a match with the Tel-to-IP call. Only if a match is not
found is a Proxy used.
Determines whether to use the device's internal routing table to
obtain the URI host name and optionally, an IP profile (per call),
even if a Proxy server is used.
ƒ
[0] Disable = Don't use internal routing table (default).
ƒ
[1] Enable = Use the 'Tel to IP Routing' table (or 'Outbound IP
Routing' table if EnableSBC is set to 1).
Notes:
Always Use Proxy
[AlwaysSendToProxy]
ƒ
This parameter appears only if the 'Use Default Proxy'
parameter is enabled.
ƒ
The domain name is used instead of a Proxy name or IP
address in the INVITE SIP URI.
Determines whether the device sends SIP messages and
responses through a Proxy server.
ƒ
[0] Disable = Use standard SIP routing rules (default).
ƒ
[1] Enable = All SIP messages and responses are sent to a
Proxy server.
Note: Applicable only if Proxy server is used (i.e., the parameter
IsProxyUsed is set to 1).
Redundant Routing Mode
[RedundantRoutingMode]
SIP ReRouting Mode
[SIPReroutingMode]
Version 5.6
Determines the type of redundant routing mechanism to implement
when a call can’t be completed using the main route.
ƒ
[0] Disable = No redundant routing is used. If the call can’t be
completed using the main route (using the active Proxy or the
first matching rule in the internal routing table), the call is
disconnected.
ƒ
[1] Routing Table = Internal routing table is used to locate a
redundant route (default).
ƒ
[2] Proxy = Proxy list is used to locate a redundant route.
Determines the routing mode after a call redirection (i.e., a 3xx SIP
response is received) or transfer (i.e., a SIP REFER request is
received).
ƒ
[0] Standard = INVITE messages that are generated as a result
of Transfer or Redirect are sent directly to the URI, according to
the Refer-To header in the REFER message or Contact header
in the 3xx response (default).
ƒ
[1] Proxy = Sends a new INVITE to the Proxy. Note: Applicable
only if a Proxy server is used and the parameter
AlwaysSendtoProxy is set to 0.
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Parameter
Description
ƒ
[2] Routing Table = Uses the Routing table to locate the
destination and then sends a new INVITE to this destination.
Notes:
ƒ
When this parameter is set to [1] and the INVITE sent to the
Proxy fails, the device re-routes the call according to the
Standard mode [0].
ƒ
When this parameter is set to [2] and the INVITE fails, the
device re-routes the call according to the Standard mode [0]. If
DNS resolution fails, the device attempts to route the call to the
Proxy. If routing to the Proxy also fails, the Redirect / Transfer
request is rejected.
ƒ
When this parameter is set to [2], the XferPrefix parameter can
be used to define different routing rules for redirected calls.
ƒ
This parameter is disregarded if the parameter
AlwaysSendToProxy is set to 1.
Proxy / Registrar Registration Parameters
(Note: The proxy and registrar parameter fields appear only if 'Enable Registration' is enabled.)
Enable Registration
[IsRegisterNeeded]
Enables the device to register to a Proxy / Registrar server.
ƒ
[0] Disable = device doesn't register to Proxy / Registrar
(default) server.
ƒ
[1] Enable = device registers to Proxy / Registrar server when
the device is powered up and at every user-defined interval
(configured by the parameter RegistrationTime).
Note: The device sends a REGISTER request for each channel or
for the entire device (according to the AuthenticationMode
parameter).
Registrar Name
[RegistrarName]
Registrar domain name. If specified, the name is used as the
Request-URI in REGISTER messages. If it isn't specified (default),
the Registrar IP address, or Proxy name or IP address is used
instead.
The valid range is up to 49 characters.
Registrar IP Address
[RegistrarIP]
The IP address (or FQDN) and optionally, port number of the SIP
Registrar server. The IP address is in dotted-decimal notation, e.g.,
201.10.8.1:<5080>.
Notes:
SIP User's Manual
ƒ
If not specified, the REGISTER request is sent to the primary
Proxy server.
ƒ
When a port number is specified, DNS NAPTR/SRV queries
aren't performed, even if DNSQueryType is set to 1 or 2.
ƒ
If the RegistrarIP is set to an FQDN and is resolved to multiple
addresses, the device also provides real-time switching
(hotswap mode) between different Registrar IP addresses
(IsProxyHotSwap is set to 1). If the first Registrar doesn't
respond to the REGISTER message, the same REGISTER
message is sent immediately to the next Proxy.
EnableProxyKeepAlive must be set to 0 for this logic to apply.
ƒ
When a specific Transport Type is defined using
RegistrarTransportType, a DNS NAPTR query is not performed
even if DNSQueryType is set to 2.
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Parameter
Registrar Transport Type
[RegistrarTransportType]
Description
Determines the transport layer used for outgoing SIP dialogs
initiated by the device to the Registrar.
ƒ
[-1] Not Configured (default)
ƒ
[0] UDP
ƒ
[1] TCP
ƒ
[2] TLS
Note: When set to ‘Not Configured’, the value of the parameter
SIPTransportType is used.
Registration Time
[RegistrationTime]
Defines the time interval (in seconds) for registering to a Proxy
server. The value is used in the Expires header. In addition, this
parameter defines the time interval between Keep-Alive messages
when the parameter EnableProxyKeepAlive is set to 2
(REGISTER).
Typically, the device registers every 3,600 sec (i.e., one hour). The
device resumes registration according to the parameter
RegistrationTimeDivider.
The valid range is 10 to 2,000,000. The default value is 180.
Re-registration Timing [%]
[RegistrationTimeDivider]
Defines the re-registration timing (in percentage). The timing is a
percentage of the re-register timing set by the Registrar server.
The valid range is 50 to 100. The default value is 50.
For example: If RegistrationTimeDivider is 70% and Registration
Expires time is 3600, the device re-sends its registration request
after 3600 x 70% = 2520 sec.
Note: This parameter may be overriden if the parameter
RegistrationTimeThreshold is greater than 0 (refer to the
description of RegistrationTimeThreshold).
Registration Retry Time
[RegistrationRetryTime]
Defines the time interval (in seconds) after which a Registration
request is resent if registration fails with a 4xx response or if there
is no response from the Proxy/Registrar server.
The default is 30 seconds. The range is 10 to 3600.
Registration Time Threshold
[RegistrationTimeThreshold]
Defines a threshold (in seconds) for re-registration timing. If this
parameter is greater than 0, but lower than the computed reregistration timing (according to the parameter
RegistrationTimeDivider), the re-registration timing is set to the
following: timing set by the Registration server in the Expires
header minus the value of the parameter
RegistrationTimeThreshold.
The valid range is 0 to 2,000,000. The default value is 0.
Re-register On INVITE Failure
[RegisterOnInviteFailure]
Enables immediate re-registration if a failure response is received
for an INVITE request sent by the device.
ReRegister On Connection
Failure
[ReRegisterOnConnectionFai
lure]
Version 5.6
ƒ
[0] Disable = Disabled (default)
ƒ
[1] Enable = Enabled
Enables the device to perform SIP Re-Registration upon TCP/TLS
connection failure.
ƒ
[0] Disable (default).
ƒ
[1] Enable.
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Parameter
Description
Miscellaneous parameters
Gateway Name
[SIPGatewayName]
Assigns a name to the device (e.g., 'gateway1.com'). Ensure that
the name you choose is the one with which the Proxy is configured
to identify the device.
Note: If specified, the device name is used as the host part of the
SIP URI in the From header. If not specified, the device's IP
address is used instead (default).
Gateway Registration Name
[GWRegistrationName]
Defines the user name that is used in the From and To headers in
REGISTER messages. If no value is specified (default) for this
parameter, the UserName parameter is used instead.
Note: This parameter is applicable only for single registration per
device (i.e., AuthenticationMode is set to 1). When the device
registers each channel separately (i.e., AuthenticationMode is set
to 0), the user name is set to the channel's phone number.
DNS Query Type
[DNSQueryType]
Enables the use of DNS Naming Authority Pointer (NAPTR) and
Service Record (SRV) queries to resolve Proxy and Registrar
servers and to resolve all domain names that appear in the Contact
and Record-Route headers.
ƒ
[0] A-Record = A-Record (default)
ƒ
[1] SRV = SRV
ƒ
[2] NAPTR = NAPTR
If set to A-Record [0], no NAPTR or SRV queries are performed.
If set to SRV [1] and the Proxy / Registrar IP address parameter,
Contact / Record-Route headers, or IP address defined in the
Routing tables contains a domain name, an SRV query is
performed. The device uses the first host name received from the
SRV query. The device then performs a DNS A-record query for
the host name to locate an IP address.
If set to NAPTR [2], an NAPTR query is performed. If it is
successful, an SRV query is sent according to the information
received in the NAPTR response. If the NAPTR query fails, an
SRV query is performed according to the configured transport type.
If the Proxy / Registrar IP address parameter, the domain name in
the Contact / Record-Route headers, or the IP address defined in
the Routing tables contains a domain name with port definition, the
device performs a regular DNS A-record query.
If a specific Transport Type is defined, a NAPTR query is not
performed.
Note: To enable NAPTR/SRV queries for Proxy servers only, use
the parameter ProxyDNSQueryType.
Proxy DNS Query Type
[ProxyDNSQueryType]
Enables the use of DNS Naming Authority Pointer (NAPTR) and
Service Record (SRV) queries to discover Proxy servers.
ƒ
[0] A-Record = A-Record (default)
ƒ
[1] SRV = SRV
ƒ
[2] NAPTR = NAPTR
If set to A-Record [0], no NAPTR or SRV queries are performed.
If set to SRV [1] and the Proxy IP address parameter contains a
domain name without port definition (e.g., ProxyIP = domain.com),
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Parameter
Description
an SRV query is performed. The SRV query returns up to four
Proxy host names and their weights. The device then performs
DNS A-record queries for each Proxy host name (according to the
received weights) to locate up to four Proxy IP addresses.
Therefore, if the first SRV query returns two domain names, and
the A-record queries return two IP addresses each, no additional
searches are performed.
If set to NAPTR [2], an NAPTR query is performed. If it is
successful, an SRV query is sent according to the information
received in the NAPTR response. If the NAPTR query fails, an
SRV query is performed according to the configured transport type.
If the Proxy IP address parameter contains a domain name with
port definition (e.g., ProxyIP = domain.com:5080), the device
performs a regular DNS A-record query.
If a specific Transport Type is defined, a NAPTR query is not
performed.
Note: When enabled, NAPTR/SRV queries are used to discover
Proxy servers even if the parameter DNSQueryType is disabled.
Number of RTX Before HotSwap
[HotSwapRtx]
Number of retransmitted INVITE/REGISTER messages before the
call is routed (hot swap) to another Proxy/Registrar.
The valid range is 1 to 30. The default value is 3.
Note: This parameter is also used for alternative routing using the
'Tel to IP Routing' table (or 'Outbound IP Routing' table if
EnableSBC is set to 1). If a domain name in the table is resolved
into two IP addresses, and if there is no response for HotSwapRtx
retransmissions to the INVITE message that is sent to the first IP
address, the device immediately initiates a call to the second IP
address.
Use Gateway Name for
OPTIONS
[UseGatewayNameForOption
s]
Determines whether the device uses its IP address or gateway
name in keep-alive SIP OPTIONS messages.
ƒ
[0] No = Use the device's IP address in keep-alive OPTIONS
messages (default).
ƒ
[1] Yes = Use 'Gateway Name' (SIPGatewayName) in keepalive OPTIONS messages.
The OPTIONS Request-URI host part contains either the device's
IP address or a string defined by the parameter SIPGatewayName.
The device uses the OPTIONS request as a keep-alive message to
its primary and redundant Proxies (i.e., the parameter
EnableProxyKeepAlive is set to 1).
User Name
[UserName]
User name used for Registration and Basic/Digest authentication
with a Proxy / Registrar server.
The parameter doesn't have a default value (empty string).
Note: Applicable only if single device registration is used (i.e.,
Authentication Mode is set to Authentication Per gateway).
Password
[Password]
Version 5.6
The password used for Basic/Digest authentication with a Proxy /
Registrar server. A single password is used for all device ports.
The default is 'Default_Passwd'.
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Parameter
Description
Cnonce
[Cnonce]
Cnonce string used by the SIP server and client to provide mutual
authentication. (Free format, i.e., 'Cnonce = 0a4f113b'). The default
is 'Default_Cnonce'.
Authentication Mode
[AuthenticationMode]
Determines the device's registration and authentication method.
ƒ
[0] Per Endpoint = Registration and Authentication separately
for each B-channel.
ƒ
[1] Per Gateway = Single Registration and Authentication for
the entire device (default).
Single Registration and Authentication (Authentication Mode = 1) is
usually defined for and digital modules.
Set Out-Of-Service On
Registration Failure
[OOSOnRegistrationFail]
Enables setting a , trunk, or the entire device (i.e., all endpoints) to
out-of-service if registration fails.
ƒ
[0] Disable = Disabled (default).
ƒ
[1] Enable = Enabled.
If the registration is per Endpoint (i.e., AuthenticationMode is set to
0) or Account (refer to ''Configuring the Trunk Group Settings'' on
page 197) and a specific endpoint/Account registration fails (SIP
4xx or no response), then that endpoint is set to out-of-service until
a success response is received in a subsequent registration
request. When the registration is per the entire device (i.e.,
AuthenticationMode is set to 1) and registration fails, all endpoints
are set to out-of-service. If all the Accounts of a specific Trunk
Group fail registration and if the Trunk Group comprises a
complete trunk, then the trunk is set to out-of-service.
Challenge Caching Mode
[SIPChallengeCachingMode]
Determines the mode for Challenge Caching, which reduces the
number of SIP messages transmitted through the network. The first
request to the Proxy is sent without authorization. The Proxy sends
a 401/407 response with a challenge. This response is saved for
further uses. A new request is resent with the appropriate
credentials. Subsequent requests to the Proxy are automatically
sent with credentials (calculated from the saved challenge). If the
Proxy doesn't accept the new request and sends another
challenge, the old challenge is replaced with the new one.
ƒ
[0] None = Challenges are not cached. Every new request is
sent without preliminary authorization. If the request is
challenged, a new request with authorization data is sent
(default)
ƒ
[1] INVITE Only = Challenges issued for INVITE requests are
cached. This prevents a mixture of REGISTER and INVITE
authorizations.
ƒ
[2] Full = Caches all challenges from the proxies.
Note: Challenge Caching is used with all proxies and not only with
the active one.
Mutual Authentication Mode
[MutualAuthenticationMode]
SIP User's Manual
Determines the device's mode of operation when Authentication
and Key Agreement (AKA) Digest Authentication is used.
ƒ
[0] Optional = Incoming requests that don't include AKA
authentication information are accepted (default).
ƒ
[1] Mandatory = Incoming requests that don't include AKA
authentication information are rejected.
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3.4.7.1.3 Proxy Sets Table
The 'Proxy Sets Table' page allows you to define Proxy Sets. A Proxy Set is a group of
Proxy servers defined by IP address or fully qualified domain name (FQDN). You can
define up to six Proxy Sets, each having a unique ID number and each containing up to five
Proxy server addresses. For each Proxy server address, you can define the transport type
(i.e., UDP, TCP, or TLS). In addition, Proxy load balancing and redundancy mechanisms
can be applied per Proxy Set (if a Proxy Set contains more than one Proxy address).
Proxy Sets can later be assigned to IP Groups of type SERVER only (refer to ''Configuring
the IP Groups'' on page 201). When the device sends an INVITE message to an IP Group,
it is sent to the IP address/domain name defined for the Proxy Set that is associated with
the specific IP Group. In other words, the Proxy Set represents the destination of the call.
Typically, for IP-to-IP call routing, at least two Proxy Sets are defined for call destination –
one for each leg (IP Group) of the call (i.e., both directions). For example, one Proxy Set for
the Internet Telephony Service provider (ITSP) interfacing with one 'leg' of the device and
another Proxy Set for the second SIP entity (e.g., ITSP) interfacing with the other 'leg' of the
device.
Note: You can also configure the Proxy Sets table using the ini file table parameters
ProxyIP and ProxySet (refer to ''SIP Configuration Parameters'' on page 284).
¾ To add Proxy servers and configure Proxy parameters, take these
5 steps:
1.
Open the 'Proxy Sets Table' page (Configuration tab > Protocol Configuration
menu > Protocol Definition submenu > Proxy Sets Table page item).
Figure 3-61: Proxy Sets Table Page
2.
From the Proxy Set ID drop-down list, select an ID for the desired group.
3.
Configure the Proxy parameters according to the following table.
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4.
Click the Submit button to save your changes.
5.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-32: Proxy Sets Table Parameters
Parameter
Proxy Set ID
Description
The Proxy Set identification number.
The valid range is 0 to 5 (i.e., up to 6 Proxy Set ID's can be
configured). The Proxy Set ID #0 is used as the default Proxy Set,
and if defined is backward compatible to the list of Proxies from
earlier releases.
Note: Although not recommended, you can use both default Proxy
Set (ID #0) and IP Groups for call routing. For example, on the
'Trunk Group Settings' page (refer to ''Configuring the Trunk Group
Settings'' on page 197), you can configure a Serving IP Group to
where you want to route specific Trunk Group's channels, while all
other device channels use the default Proxy Set. At the same, you
can also use IP Groups in the 'Tel to IP Routing' table (refer to ''Tel
to IP Routing Table'' on page 175) or 'Outbound IP Routing' table if
EnableSBC is set to 1 (refer to “Outbound IP Routing Table” on
page 178) to configure the default Proxy Set if the parameter
PreferRouteTable is set to 1.
To summarize, if the default Proxy Set is used, the INVITE
message is sent according to the following preferences:
ƒ
To the Trunk Group's Serving IP Group ID, as defined in the
'Trunk Group Settings' table.
ƒ
According to the 'Tel to IP Routing' table (or 'Outbound IP
Routing' table if EnableSBC is set to 1), if the parameter
PreferRouteTable is set to 1.
ƒ
To the default Proxy.
Typically, when IP Groups are used, there is no need to use the
default Proxy, and all routing and registration rules can be
configured using IP Groups and the Account tables (refer to
''Configuring the Account Table'' on page 204).
Proxy Address
SIP User's Manual
The IP address (and optionally port number) of the Proxy server.
Up to five IP addresses can be configured per Proxy Set. Enter the
IP address as an FQDN or in dotted-decimal notation (e.g.,
201.10.8.1). You can also specify the selected port in the format:
<IP address>:<port>.
If you enable Proxy Redundancy (by setting the parameter
EnableProxyKeepAlive to 1 or 2), the device can operate with
multiple Proxy servers. If there is no response from the first
(primary) Proxy defined in the list, the device attempts to
communicate with the other (redundant) Proxies in the list. When a
redundant Proxy is located, the device either continues operating
with it until the next failure occurs, or reverts to the primary Proxy
(refer to the parameter ProxyRedundancyMode). If none of the
Proxy servers respond, the device goes over the list again.
The device also provides real-time switching (Hot-Swap mode)
between the primary and redundant proxies (refer to the parameter
IsProxyHotSwap). If the first Proxy doesn't respond to the INVITE
message, the same INVITE message is immediately sent to the
next Proxy in the list. The same logic applies to REGISTER
messages (if RegistrarIP is not defined).
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Parameter
Description
Notes:
Transport Type
ƒ
If EnableProxyKeepAlive is set to 1 or 2, the device monitors
the connection with the Proxies by using keep-alive messages
(OPTIONS or REGISTER).
ƒ
To use Proxy Redundancy, you must specify one or more
redundant Proxies.
ƒ
When a port number is specified (e.g., domain.com:5080), DNS
NAPTR/SRV queries aren't performed, even if
ProxyDNSQueryType is set to 1 or 2.
The transport type per Proxy server.
ƒ
[0] UDP
ƒ
[1] TCP
ƒ
[2] TLS
ƒ
[-1] = Undefined
Note: If no transport type is selected, the value of the global
parameter SIPTransportType is used (refer to ''SIP General
Parameters'' on page 121).
Proxy Load Balancing Method
[ProxyLoadBalancingMethod]
Enables the Proxy Load Balancing mechanism per Proxy Set ID.
ƒ
[0] Disable = Load Balancing is disabled (default).
ƒ
[1] Round Robin = Round Robin.
ƒ
[2] Random Weights = Random Weights.
When the Round Robin algorithm is used, a list of all possible
Proxy IP addresses is compiled. This list includes all IP addresses
per Proxy Set, after necessary DNS resolutions (including NAPTR
and SRV, if configured). After this list is compiled, the Proxy KeepAlive mechanism (according to parameters EnableProxyKeepAlive
and ProxyKeepAliveTime) tags each entry as 'offline' or 'online'.
Load balancing is only performed on Proxy servers that are tagged
as 'online'.
All outgoing messages are equally distributed across the list of IP
addresses. REGISTER messages are also distributed unless a
RegistrarIP is configured.
The IP addresses list is refreshed according to
ProxyIPListRefreshTime. If a change in the order of the entries in
the list occurs, all load statistics are erased and balancing starts
over again.
When the Random Weights algorithm is used, the outgoing
requests are not distributed equally among the Proxies. The
weights are received from the DNS server by using SRV records.
The device sends the requests in such a fashion that each Proxy
receives a percentage of the requests according to its' assigned
weight. A single FQDN should be configured as a Proxy IP
address. The Random Weights Load Balancing is not used in the
following scenarios:
Version 5.6
ƒ
The Proxy Set includes more than one Proxy IP address.
ƒ
The only Proxy defined is an IP address and not an FQDN.
ƒ
SRV is not enabled (DNSQueryType).
ƒ
The SRV response includes several records with a different
Priority value.
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Parameter
Enable Proxy Keep Alive
[EnableProxyKeepAlive]
Description
Determines whether Keep-Alive with the Proxy is enabled or
disabled. This parameter is configured per Proxy Set.
ƒ
[0] Disable = Disable (default).
ƒ
[1] Using OPTIONS = Enables Keep-Alive with Proxy using
OPTIONS.
ƒ
[2] Using REGISTER = Enable Keep-Alive with Proxy using
REGISTER.
If set to 'Using OPTIONS', the SIP OPTIONS message is sent
every user-defined interval, as configured by the parameter
ProxyKeepAliveTime. If set to 'Using REGISTER', the SIP
REGISTER message is sent every user-defined interval, as
configured by the parameter RegistrationTime. Any response from
the Proxy, either success (200 OK) or failure (4xx response) is
considered as if the Proxy is communicating correctly.
Notes:
Proxy Keep Alive Time
[ProxyKeepAliveTime]
ƒ
For Survivability mode for USER-type IP Groups, this
parameter must be enabled (1 or 2).
ƒ
This parameter must be set to 'Using OPTIONS' when Proxy
redundancy is used.
ƒ
When this parameter is set to 'Using REGISTER', the homing
redundancy mode is disabled.
ƒ
When the active proxy doesn't respond to INVITE messages
sent by the device, the proxy is tagged as 'offline'. The behavior
is similar to a Keep-Alive (OPTIONS or REGISTER) failure.
Defines the Proxy keep-alive time interval (in seconds) between
Keep-Alive messages. This parameter is configured per Proxy Set.
The valid range is 5 to 2,000,000. The default value is 60.
Note: This parameter is applicable only if the parameter
EnableProxyKeepAlive is set to 1 (OPTIONS). When the
parameter EnableProxyKeepAlive is set to 2 (REGISTER), the
time interval between Keep-Alive messages is determined by the
parameter RegistrationTime.
Is Proxy Hot-Swap
[IsProxyHotSwap]
Enables the Proxy Hot-Swap redundancy mode per Proxy Set.
ƒ
[0] No = Disabled (default).
ƒ
[1] Yes = Proxy Hot-Swap mode is enabled.
If Proxy Hot-Swap is enabled, the SIP INVITE/REGISTER
message is initially sent to the first Proxy/Registrar server. If there
is no response from the first Proxy/Registrar server after a specific
number of retransmissions (configured by the parameter
HotSwapRtx), the INVITE/REGISTER message is resent to the
next redundant Proxy/Registrar server.
3.4.7.1.4 Coders
The 'Coders' page allows you to configure up to five coders (and their attributes) for the
device. The first coder in the list is the highest priority coder and is used by the device
whenever possible. If the far-end device cannot use the first coder, the device attempts to
use the next coder in the list, and so forth.
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Notes:
•
The device always uses the packetization time requested by the remote
side for sending RTP packets.
•
For an explanation on V.152 support (and implementation of T.38 and
VBD coders), refer to ''Supporting V.152 Implementation'' on page 357.
•
You can also configure the Coders table using the ini file table parameter
CoderName (refer to ''SIP Configuration Parameters'' on page 284).
The coders supported by the device are listed in the table below:
Table 3-33: Supported Coders
Coder Name
Packetization
Time
Rate
G.711 A-law
[g711Alaw64k]
10, 20 (default),
30, 40, 50, 60, 80,
100, 120
Always 64
G.711 U-law
[g711Ulaw64k]
10, 20 (default),
30, 40, 50, 60, 80,
100, 120
Always 64
EG.711 A-law
[eg711Alaw]
10 (default), 20, 30
Always 64
Dynamic (0120)
N/A
EG.711 U-law
[eg711Ulaw]
10 (default), 20, 30
Always 64
Dynamic (0120)
N/A
G.729
[g729]
10, 20 (default),
30, 40, 50, 60, 80,
100
Always 8
Always 18
ƒ
Disable [0]
ƒ
Enable [1]
ƒ
Enable w/o
Adaptations [2]
G.723.1
[g7231]
30 (default), 60,
90, 120
5.3 [0], 6.3
[1]
(default)
Always 4
ƒ
Disable [0]
ƒ
Enable [1]
G.726
[g726]
10, 20 (default),
30, 40, 50, 60, 80,
100, 120
16 [0], 24
[1], 32 [2]
(default)
40 [3]
Dynamic (0120)
ƒ
Disable [0]
ƒ
Enable [1]
GSM-FR
[gsmFullRate]
20 (default), 40,
60, 80
Always 13
Always 3
ƒ
Disable [0]
ƒ
Enable [1]
GSM-EFR
[gsmEnhancedFullRate]
0, 20 (default), 30,
40, 50, 60, 80, 100
12.2
Dynamic (0120)
ƒ
Disable [0]
ƒ
Enable [1]
AMR
[Amr]
20 (default)
4.75 [0],
5.15 [1],
5.90 [2],
6.70 [3],
7.40 [4],
7.95 [5],
10.2 [6],
12.2 [7]
(default)
Dynamic (0120)
ƒ
Disable [0]
ƒ
Enable [1]
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Payload
Type
Always 8
Always 0
Silence
Suppression
ƒ
Disable [0]
ƒ
Enable [1]
ƒ
Disable [0]
ƒ
Enable [1]
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Coder Name
Packetization
Time
Rate
Payload
Type
EVRC
[Evrc]
20 (default), 40,60,
80, 100
Variable
[0]
(default),
1/8 [1], 1/2
[3], Full [4]
Dynamic (0120)
ƒ
Disable [0]
ƒ
Enable [1]
iLBC
[iLBC]
20 (default), 40,
60, 80, 100, 120
15
(default)
Dynamic (0120)
ƒ
Disable [0]
ƒ
Enable [1]
30 (default), 60,
90, 120
13
MS-GSM
[gsmMS]
40 (default)
Always 13
Always 3
ƒ
Disable [0]
ƒ
Enable [1]
QCELP
[QCELP]
20 (default), 40,
60, 80, 100, 120
Always 13
ƒ
Disable [0]
ƒ
Enable [1]
Transparent
[Transparent]
20 (default), 40,
60, 80, 100, 120
Always 64
Dynamic (0120)
ƒ
Disable [0]
ƒ
Enable [1]
G.711A-law_VBD
[g711AlawVbd]
10, 20 (default),
30, 40, 50, 60, 80,
100, 120
Always 64
Dynamic (0120)
N/A
G.711U-law_VBD
[g711UlawVbd]
10, 20 (default),
30, 40, 50, 60, 80,
100, 120
Always 64
Dynamic (0120)
N/A
T.38
[t38fax]
N/A
N/A
N/A
N/A
Always 12
Silence
Suppression
¾ To configure the device's coders, take these 9 steps:
1.
Open the 'Coders' page (Configuration tab > Protocol Configuration menu >
Protocol Definition submenu > Coders page item).
Figure 3-62: Coders Page
2.
From the 'Coder Name' drop-down list, select the coder you want to use. For the full
list of available coders and their corresponding attributes, refer to the table below.
3.
From the 'Packetization Time' drop-down list, select the packetization time (in msec)
for the selected coder. The packetization time determines how many coder payloads
are combined into a single RTP packet.
4.
From the 'Rate' drop-down list, select the bit rate (in kbps) for the selected coder.
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5.
In the 'Payload Type' field, if the payload type for the selected coder is dynamic, enter
a value from 0 to 120 (payload types of 'well-known' coders cannot be modified). The
payload type identifies the format of the RTP payload.
6.
From the 'Silence Suppression' drop-down list, enable or disable the silence
suppression option for the selected coder.
7.
Repeat steps 2 through 6 for the second to fifth optional coders.
8.
Click the Submit button to save your changes.
9.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Notes:
Version 5.6
•
Each coder (i.e., 'Coder Name') can appear only once.
•
If packetization time and / or rate are not specified, the default value is
applied.
•
Only the packetization time of the first coder in the coder list is declared
in INVITE / 200 OK SDP, even if multiple coders are defined.
•
For G.729, it's also possible to select silence suppression without
adaptations.
•
If the coder G.729 is selected and silence suppression is disabled (for
this coder), the device includes the string 'annexb=no' in the SDP of the
relevant SIP messages. If silence suppression is enabled or set to
'Enable w/o Adaptations', 'annexb=yes' is included. An exception to this
logic is when the remote gateway is a Cisco device (IsCiscoSCEMode).
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3.4.7.1.5 DTMF & Dialing Parameters
The 'DTMF & Dialing' page is used to configure parameters associated with dual-tone multifrequency (DTMF) and dialing.
¾ To configure the DTMF and dialing parameters, take these 4 steps:
1.
Open the 'DTMF & Dialing' page (Configuration tab > Protocol Configuration menu
> Protocol Definition submenu > DTMF & Dialing page item).
Figure 3-63: DTMF & Dialing Page
2.
Configure the DTMF and dialing parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-34: DTMF and Dialing Parameters
Parameter
Description
Max Digits in Phone Num
[MaxDigits]
Defines the maximum number of collected destination number digits
that can be received from the Tel side when Tel-to-IP ISDN overlap
dialing is performed . When the number of collected digits reaches the
maximum, the device uses these digits for the called destination
number.
The valid range is 1 to 49. The default value is 30.
Note: Digit Mapping Rules can be used instead.
Inter Digit Timeout for
Overlap Dialing [sec]
[TimeBetweenDigits]
Defines the time (in seconds) that the device waits between digits that
are received from the Tel side when Tel-to-IP overlap dialing is
performed (ISDN uses overlap dialing). When this inter-digit timeout
expires, the device uses the collected digits to dial the called
destination number.
The valid range is 1 to 10. The default value is 4.
Declare RFC 2833 in SDP
[RxDTMFOption]
Defines the supported Receive DTMF negotiation method.
SIP User's Manual
ƒ
[0] No = Don't declare RFC 2833 telephony-event parameter in
SDP.
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Parameter
Description
ƒ
[3] Yes = Declare RFC 2833 telephony-event parameter in SDP
(default).
The device is designed to always be receptive to RFC 2833 DTMF
relay packets. Therefore, it is always correct to include the 'telephonyevent' parameter as default in the SDP. However, some devices use
the absence of the 'telephony-event' in the SDP to decide to send
DTMF digits in-band using G.711 coder. If this is the case, you can set
RxDTMFOption to 0.
1st to 5th Tx DTMF Option
[TxDTMFOption]
Determines a single or several preferred transmit DTMF negotiation
methods.
ƒ
[0] Not Supported = No negotiation - DTMF digits are sent
according to the parameters DTMFTransportType and
RFC2833PayloadType (default).
ƒ
[1] INFO (Nortel) = Sends DTMF digits according to IETF <draftchoudhuri-sip-info-digit-00>.
ƒ
[2] NOTIFY = Sends DTMF digits according to <draft-mahysipping-signaled-digits-01>.
ƒ
[3] INFO (Cisco) = Sends DTMF digits according to Cisco format.
ƒ
[4] RFC 2833.
ƒ
[5] INFO (Korea) = Sends DTMF digits according to Korea
Telecom format.
Notes:
Version 5.6
ƒ
DTMF negotiation methods are prioritized according to the order of
their appearance.
ƒ
When out-of-band DTMF transfer is used ([1], [2], [3], or [5]), the
parameter DTMFTransportType is automatically set to 0 (DTMF
digits are erased from the RTP stream).
ƒ
When RFC 2833 (4) is selected, the device:
1) Negotiates RFC 2833 Payload Type (PT) using local and remote
SDPs.
2) Sends DTMF packets using RFC 2833 PT according to the PT in
the received SDP.
3) Expects to receive RFC 2833 packets with the same PT as
configured by the parameter RFC2833PayloadType.
4) Sends DTMF digits in transparent mode (as part of the voice
stream).
ƒ
When TxDTMFOption is set to 0, the RFC 2833 PT is set according
to the parameter RFC2833PayloadType for both transmit and
receive.
ƒ
The ini file table parameter TxDTMFOption can be repeated 5
times for configuring the DTMF transmit methods.
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Parameter
RFC 2833 Payload Type
[RFC2833PayloadType]
Description
The RFC 2833 DTMF relay dynamic payload type.
The valid range is 96 to 99, and 106 to 127. The default is 96. The
100, 102 to 105 range is allocated for proprietary usage.
Notes:
Hook-Flash Option
[HookFlashOption]
ƒ
Certain vendors (e.g., Cisco) use payload type 101 for RFC 2833.
ƒ
When RFC 2833 payload type (PT) negotiation is used (the
parameter TxDTMFOption is set to 4), this payload type is used for
the received DTMF packets. If negotiation isn't used, this payload
type is used for receive and for transmit.
Determines the supported hook-flash Transport Type (i.e., method by
which hook-flash is sent and received).
ƒ
[0] Not Supported = Hook-Flash indication isn't sent (default).
ƒ
[1] INFO = Send proprietary INFO message with Hook-Flash
indication.
ƒ
[4] RFC 2833
ƒ
[5] INFO (Lucent) = Send proprietary INFO message with HookFlash indication.
Notes:
Digit Mapping Rules
[DigitMapping]
ƒ
The RFC 2833 (4) option is currently not supported.
ƒ
The DTMF HookFlashCode is send to IP according to the
parameter HookFlashOption.
Digit map pattern (used to reduce the dialing period when Overlap
dialing is used). If the digit string (i.e., dialed number) matches one of
the patterns in the digit map, the device stops collecting digits and
establishes a call with the collected number.
The digit map pattern can contain up to 52 options, each separated by
a vertical bar (|). The maximum length of the entire digit pattern is 152
characters.
Available notations:
ƒ
[n-m]: Range of numbers (not letters).
ƒ
. (single dot): Repeat digits until next notation (e.g., T).
ƒ
x: Any single digit.
ƒ
T: Dial timeout (configured by the parameter TimeBetweenDigits).
ƒ
S: Immediately applies a specific rule that is part of a general rule.
For example, if your digit map includes a general rule 'x.T' and a
specific rule '11x', for the specific rule to take precedence over the
general rule, append 'S' to the specific rule (i.e., '11xS').
An example of a digit map is shown below:
11xS|00T|[1-7]xxx|8xxxxxxx|#xxxxxxx|*xx|91xxxxxxxxxx|9011x.T
In the example above, the last rule can apply to International numbers
- 9 for dialing tone, 011 Country Code, and then any number of digits
for the local number ('x.').
Note: For PRI interfaces, the digitmap mechanism is applicable only
when ISDN Overlap dialing is used (ISDNRxOverlap is set to 1).
Dial Tone Duration [sec]
[TimeForDialTone]
SIP User's Manual
Duration (in seconds) that the dial tone is played to an ISDN terminal.
This parameter is applicable for overlap dialing when
ISDNInCallsBehavior = 65536. The dial tone is played if the ISDN
SETUP message doesn't include the called number.
The valid range is 0 to 60. The default is 5.
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Parameter
Description
Default Destination Number
[DefaultNumber]
Defines the default destination phone number used if the received
message doesn't contain a called party number and no phone number
is configured in the 'Trunk Group' table (refer to “Configuring the Trunk
Group Table” on page 195). The parameter is used as a starting
number for the list of channels comprising all trunk groups in the
device.
The default value is 1000.
Special Digit
Representation
[UseDigitForSpecialDTMF]
Defines the representation for ‘special’ digits (‘*’ and ‘#’) that are used
for out-of-band DTMF signaling (using SIP INFO/NOTIFY).
3.4.7.2
ƒ
[0] Special = Uses the strings ‘*’ and ‘#’ (default).
ƒ
[1] Numeric = Uses the numerical values 10 and 11.
Configuring the SIP Advanced Parameters
The SIP Advanced Parameters submenu allows you to configure advanced SIP control
protocol parameters. This submenu contains the following page items:
„
Advanced Parameters (refer to ''General Parameters'' on page 151)
„
Supplementary Services (refer to ''Supplementary Services'' on page 159)
„
Stand-Alone Survivability (refer to “Stand-Alone Survivability” on page 161)
„
SBC Configuration (refer to “SBC Configuration” on page 163)
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3.4.7.2.1 Advanced Parameters
The 'Advanced Parameters' page allows you to configure general control protocol
parameters.
¾ To configure the advanced general protocol parameters, take these
4 steps:
1.
Open the 'Advanced Parameters' page (Configuration tab > Protocol Configuration
menu > SIP Advanced Parameters submenu > Advanced Parameters page item).
Figure 3-64: Advanced Parameters Page
2.
Configure the parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
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Table 3-35: Advanced Parameters Description
Parameter
Description
General
IP Security
[SecureCallsFromIP]
Determines whether the device accepts SIP calls received from only IP
addresses defined in the 'Tel to IP Routing' table (refer to ''Tel to IP
Routing Table'' on page 175) or 'Outbound IP Routing' table if
EnableSBC is set to 1 (refer to “Outbound IP Routing Table” on page
178). This is useful in preventing unwanted SIP calls or messages
and/or VoIP spam.
ƒ
[0] Disable = device accepts all SIP calls (default).
ƒ
[1] Enable = device accepts SIP calls only from IP addresses
defined in the 'Tel to IP Routing' table (or 'Outbound IP Routing'
table). The device rejects all calls from unknown IP addresses.
Note: Specifying the IP address of a Proxy server in the 'Tel to IP
Routing' table (or 'Outbound IP Routing' table) enables the device to
accept only calls originating from the Proxy server while rejecting all
other calls that don’t appear in this table.
Filter Calls to IP
[FilterCalls2IP]
Enables filtering of Tel-to-IP calls when a Proxy is used (i.e.,
IsProxyUsed parameter is set to 1 -- refer to ''Proxy & Registration
Parameters'' on page 132).
ƒ
[0] Don't Filter = device doesn't filter calls when using a Proxy.
(default)
ƒ
[1] Filter = Filtering is enabled.
When this parameter is enabled and a Proxy is used, the device first
checks the 'Tel-to-IP Routing' table or 'Outbound IP Routing' table
before making a call through the Proxy. If the number is not allowed
(i.e., number isn't listed in the table or a call restriction routing rule of IP
address 0.0.0.0 is applied), the call is released.
Note: When no Proxy is used, this parameter must be disabled and
filtering is according to the 'Tel-to-IP Routing' table (or 'Outbound IP
Routing' table).
Enable Digit Delivery to IP
[EnableDigitDelivery2IP]
The Digit Delivery feature enables sending DTMF digits to the
destination IP address after the Tel-to-IP call is answered.
ƒ
[0] Disable = Disabled (default).
ƒ
[1] Enable = Enable digit delivery to IP.
To enable this feature, modify the called number to include at least one
'p' character. The device uses the digits before the 'p' character in the
initial INVITE message. After the call is answered, the device waits for
the required time (number of 'p' multiplied by 1.5 seconds) and then
sends the rest of the DTMF digits using the method chosen (in-band or
out-of-band).
Note: The called number can include several 'p' characters (1.5
seconds pause), for example, 1001pp699, 8888p9p300.
Enable Digit Delivery to
Tel
[EnableDigitDelivery]
Version 5.6
Enables the Digit Delivery feature, which sends DTMF digits (of the
called number) to the device's B-channel (phone line) after the call is
answered [line offhooked (FXS) or seized (FXO)] for IP-to-Tel calls.
ƒ
[0] Disable = Disabled (default).
ƒ
[1] Enable = Enable Digit Delivery feature for the device (two-stage
dialing).
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Parameter
Description
If the called number in IP-to-Tel call includes the characters 'w' or 'p',
the device places a call with the first part of the called number (before
'w' or 'p') , and plays DTMF digits after the call is answered. If the
character 'w' is used, the device waits for detection of dial tone before it
starts playing DTMF digits. For example, if the called number is
'1007766p100', the device places a call with 1007766 as the destination
number, then after the call is answered, it waits 1.5 seconds ('p') and
plays the rest of the number (100) as DTMF digits.
Additional examples: 1664wpp102, 66644ppp503, and
7774w100pp200.
RTP Only Mode
[RTPOnlyMode]
Enables the device to start sending and/or receiving RTP packets to
and from remote endpoints without the need to establish a Control
session. The remote IP address is determined according to the 'Tel to
IP Routing' table (refer to ''Tel to IP Routing Table'' on page 175) or
'Outbound IP Routing' table (refer to “Outbound IP Routing Table” on
page 178). The port is the same port as the local RTP port (set by
BaseUDPPort and the channel on which the call is received).
ƒ
[0] Disable = Disable (default).
ƒ
[1] Transmit & Receive = Send and receive RTP.
ƒ
[2] Transmit Only= Send RTP only.
ƒ
[3] Receive Only= Receive RTP only.
Notes:
PSTN Alert Timeout
[PSTNAlertTimeout]
ƒ
To configure the RTP Only mode per trunk, use the
RTPOnlyModeForTrunk_ID (refer to “Configuring the Trunk Settings”
on page 82).
ƒ
If per trunk configuration (using RTPOnlyModeForTrunk) is set to
other than default, the RTPOnlyMode parameter value is overridden.
Alert Timeout (in seconds) (ISDN T301 timer) for calls to PSTN. This
timer is used between the time a SETUP message is sent to the Tel
side (IP-to-Tel call establishment) and a CONNECT message is
received. If an ALERTING message is received, the timer is restarted.
The default is 180 seconds. The range is 1 to 600.
Note: If per trunk configuration (using TrunkPSTNAlertTimeout) is set to
other than default (refer to “Configuring the Trunk Settings” on page 82),
the PSTNAlertTimeout parameter value is overridden.
Reanswer Time
[RegretTime]
Determines the time period the device waits for an MFC R2 Resume
(Reanswer) signal once a Suspend (Clear back) signal is received from
the PBX. If this timer expires, the call is released.
Note: Applicable only for MFC R2 CAS Brazil variant.
The valid range is 0 to 255 (in seconds). The default value is 0.
Disconnect and Answer Supervision
Send Digit Pattern on
Connect
[TelConnectCode]
Defines a digit pattern to send to the Tel side after SIP 200 OK is
received from the IP side. The digit pattern is a pre-defined DTMF
sequence that is used to indicate an answer signal (e.g., for billing).
The valid range is 1 to 8 characters.
Note: This parameter is applicable to FXO and CAS.
Disconnect on Broken
Connection
[DisconnectOnBrokenCo
SIP User's Manual
Determines whether the device releases the call if RTP packets are not
received within a user-defined timeout.
ƒ
[0] No
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Parameter
nnection]
Description
ƒ
[1] Yes (default)
Notes:
Broken Connection
Timeout
[BrokenConnectionEven
tTimeout]
ƒ
The timeout is set by the parameter
BrokenConnectionEventTimeout.
ƒ
This feature is applicable only if the RTP session is used without
Silence Compression. If Silence Compression is enabled, the device
doesn't detect a broken RTP connection.
ƒ
During a call, if the source IP address (from where the RTP packets
are sent) is changed without notifying the device, the device filters
these RTP packets. To overcome this, set
DisconnectOnBrokenConnection to 0; the device doesn't detect RTP
packets arriving from the original source IP address and switches
(after 300 msec) to the RTP packets arriving from the new source IP
address.
The time period (in 100 msec units) that an RTP packet is not received
after which a call is disconnected.
The valid range is 1 to 1,000. The default value is 100 (i.e., 10
seconds).
Notes:
Disconnect Call on Silence
Detection
[EnableSilenceDisconne
ct]
ƒ
Applicable only if DisconnectOnBrokenConnection = 1.
ƒ
Currently, this feature works only if Silence Suppression is disabled.
Determines whether calls are disconnected after detection of silence.
ƒ
[1] Yes = The device disconnects calls in which silence occurs (in
both call directions) for more than a user-defined time.
ƒ
[0] No = Call is not disconnected when silence is detected (default).
The silence duration can be set by the FarEndDisconnectSilencePeriod
parameter (default 120).
Note: To activate this feature, set EnableSilenceCompression and
FarEndDisconnectSilenceMethod to 1.
Silence Detection Period
[sec]
[FarEndDisconnectSilen
cePeriod]
Duration of silence period (in seconds) prior to call disconnection.
The range is 10 to 28,800 (i.e., 8 hours). The default is 120 seconds.
Silence Detection Method
[FarEndDisconnectSilen
ceMethod]
Silence detection method.
Enable Fax Re-Routing
[EnableFaxReRouting]
Note: This parameter is applicable only to devices that use DSP
templates 2 and 3.
ƒ
[0] None = Silence detection option is disabled.
ƒ
[1] Packets Count = According to packet count.
ƒ
[2] Voice/Energy Detectors = N/A.
ƒ
[3] All = N/A.
Enables or disables re-routing of Tel-to-IP calls that are identified as fax
calls.
ƒ
[0] Disable = Disabled (default).
ƒ
[1] Enable = Enabled.
If a CNG tone is detected on the Tel side of a Tel-to-IP call, a 'FAX'
prefix is appended to the destination number before routing and
manipulations. An entry of ‘FAX’ as destination number in the 'Tel-to-IP
Routing' table is then used to route the call, and the destination number
Version 5.6
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Parameter
Description
manipulation mechanism is used to remove the 'FAX' prefix, if required.
If the initial INVITE used to establish the voice call (not fax) was already
sent, a CANCEL (if not connected yet) or a BYE (if already connected)
is sent to tear down the voice call.
Notes:
ƒ
To enable this feature, set CNGDetectorMode to 2, and IsFaxUsed
to 1, 2, or 3.
ƒ
The 'FAX' prefix in routing and manipulation tables is case sensitive.
CDR and Debug
CDR Server IP Address
[CDRSyslogServerIP]
Defines the destination IP address to where CDR logs are sent.
The default value is a null string, which causes CDR messages to be
sent with all Syslog messages to the Syslog server.
Note: The CDR messages are sent to UDP port 514 (default Syslog
port).
CDR Report Level
[CDRReportLevel]
Determines whether Call Detail Records (CDR) are sent to the Syslog
server and when they are sent.
ƒ
[0] None = CDRs are not used (default).
ƒ
[1] End Call = CDR is sent to the Syslog server at the end of each
call.
ƒ
[2] Start & End Call = CDR report is sent to Syslog at the start and
end of each call.
ƒ
[3] Connect & End Call = CDR report is sent to Syslog at connection
and at the end of each call.
ƒ
[4] Start & Connect & End Call = CDR report is sent to Syslog at the
start, at connection, and at the end of each call.
The CDR Syslog message complies with RFC 3161 and is identified by:
Facility = 17 (local1) and Severity = 6 (Informational).
Debug Level
[GwDebugLevel]
Syslog debug logging level.
ƒ
[0] 0 = Debug is disabled (default).
ƒ
[1] 1 = Flow debugging is enabled.
ƒ
[2] 2 = Flow and device interface debugging are enabled.
ƒ
[3] 3 = Flow, device interface, and stack interface debugging are
enabled.
ƒ
[4] 4 = Flow, device interface, stack interface, and session manager
debugging are enabled.
ƒ
[5] 5 = Flow, device interface, stack interface, session manager, and
device interface expanded debugging are enabled.
Note: Usually set to 5 if debug traces are needed.
Misc. Parameters
Progress Indicator to IP
[ProgressIndicator2IP]
SIP User's Manual
ƒ
[-1] Not Configured = for ISDN spans, the progress indicator (PI) that
is received in ISDN Proceeding, Progress, and Alert messages is
used as described in the options below. (default)
ƒ
[0] No PI = For IP-to-Tel calls, the device sends 180 Ringing SIP
response to IP after receiving ISDN Alert or (for CAS) after placing a
call to PBX/PSTN.
ƒ
[1] PI =1, [8] PI =8: For IP-to-Tel calls, if EnableEarlyMedia = 1, the
device sends 180 Ringing with SDP in response to an ISDN Alert or
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Parameter
Description
it sends a 183 Session Progress message with SDP in response to
only the first received ISDN Proceeding or Progress message after a
call is placed to PBX/PSTN over the trunk.
Enable X-Channel Header
[XChannelHeader]
Determines whether the x-channel header is added to SIP messages
for trunk / B-channel information.
ƒ
[0] Disable = x-channel header is not used (default).
ƒ
[1] Enable = x-channel header is generated with trunk/B-channel
and IP address information.
The header provides information on the E1/T1 physical trunk/B-channel
on which the call is received or placed. For example, 'x-channel:
DS/DS1-5/22;IP=192.168.13.1', where 'DS/DS-1' is a constant string, '5'
is the trunk number, '22' is the B-channel, and in addition, the device's
IP address is added to the header. This header is generated by the
device and is sent in INVITE messages and 183/180/200OK responses.
Enable Busy Out
[EnableBusyOut]
Determines whether the Busy Out feature is enabled.
ƒ
[0] Disable = 'Busy out' feature is not used (default).
ƒ
[1] Enable = 'Busy out' feature is enabled.
When Busy Out is enabled and certain scenarios exist, the device
performs the following:
All E1/T1 trunks are automatically taken out of service by taking down
the D-Channel or by sending a Service Out message for T1 PRI trunks
supporting these messages (NI-2, 4/5-ESS, DMS-100, and Meridian).
These behaviors are performed due to one of the following scenarios:
ƒ
Physically disconnected from the network (i.e., Ethernet cable is
disconnected).
ƒ
The Ethernet cable is connected, but the device can't communicate
with any host. Note that LAN Watch-Dog must be activated
(EnableLANWatchDog = 1).
ƒ
The device can't communicate with the proxy (according to the Proxy
keep-alive mechanism) and no other alternative exists to send the
call.
ƒ
The IP Connectivity mechanism is enabled (using
AltRoutingTel2IPEnable) and there is no connectivity to any
destination IP address.
Note: The Busy Out behavior varies between different protocol types.
Default Release Cause
[DefaultReleaseCause]
Defines the default Release Cause (sent to IP) for IP-to-Tel calls when
the device initiates a call release and an explicit matching cause for this
release isn't found.
The default release cause is NO_ROUTE_TO_DESTINATION (3).
Other common values include NO_CIRCUIT_AVAILABLE (34),
DESTINATION_OUT_OF_ORDER (27), etc.
Notes:
Version 5.6
ƒ
The default release cause is described in the Q.931 notation and is
translated to corresponding SIP 40x or 50x values (e.g., 3 to SIP 404
and 34 to SIP 503).
ƒ
When the Trunk is disconnected or is not synchronized, the internal
cause is 27. This cause is mapped, by default, to SIP 502.
ƒ
For mapping SIP-to-Q.931 and Q.931-to-SIP release causes, refer to
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Parameter
Description
Release Reason Mapping on page 394.
ƒ
For a list of SIP responses-Q.931 release cause mapping, refer to
''Release Cause Mapping'' on page 189.
Delay After Reset [sec]
[GWAppDelayTime]
Defines the time interval (in seconds) that the device's operation is
delayed after a reset.
The valid range is 0 to 45. The default value is 7 seconds.
Note: This feature helps to overcome connection problems caused by
some LAN routers or IP configuration parameters' modifications by a
DHCP server.
Max Number of Active
Calls
[MaxActiveCalls]
Defines the maximum number of simultaneous active calls supported by
the device. If the maximum number of calls is reached, new calls are not
established.
The default value is the maximum available channels (no restriction on
the maximum number of calls). The valid range is 1 to 240.
Max Call Duration (min)
[MaxCallDuration]
Defines the maximum call duration (in minutes). If this time expires, both
sides of the call are released (IP and Tel).
The valid range is 0 to 35,791. The default is 0 (i.e., no limitation).
Enable LAN Watchdog
[EnableLanWatchDog]
Determines whether the LAN Watch-Dog feature is enabled.
ƒ
[0] Disable = Disable LAN Watch-Dog (default).
ƒ
[1] Enable = Enable LAN Watch-Dog.
When LAN Watch-Dog is enabled, the device's overall communication
integrity is checked periodically. If no communication for about 3
minutes is detected, the device performs a self test.
If the self test succeeds, the problem is logical link down (i.e., Ethernet
cable disconnected on the switch side), and the Busy Out mechanism is
activated if enabled (EnableBusyOut = 1).
If the self test fails, the device restarts to overcome internal fatal
communication error.
Note: Enable LAN Watchdog is relevant only if the Ethernet connection
is full duplex.
Enable User-Information
Usage
[EnableUserInfoUsage]
First Call Ringback Tone
ID
[FirstCallRBTId]
Enables or disables usage of the User Information loaded to the device
in the User Information auxiliary file. (For a description on User
Information, refer to ''Loading Auxiliary Files'' on page 231.)
ƒ
[0] Disable = Disabled (default).
ƒ
[1] Enable = Enabled.
Determines the index of the first Ringback Tone in the CPT file. This
option enables an Application server to request the device to play a
distinctive Ringback tone to the calling party according to the
destination of the call. The tone is played according to the Alert-Info
header received in the 180 Ringing SIP response (the value of the AlertInfo header is added to the value of this parameter).
The valid range is -1 to 1,000. The default value is -1 (i.e., play standard
Ringback tone).
Notes:
SIP User's Manual
ƒ
It is assumed that all Ringback Tones are defined in sequence in the
CPT file.
ƒ
In case of an MLPP call, the device uses the value of this parameter
plus 1 as the index of the Ringback tone in the CPT file (e.g., if this
value is set to 1, then the index is 2, i.e., 1 + 1).
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3.4.7.2.2 Supplementary Services
The 'Supplementary Services' page is used to configure parameters that are associated
with supplementary services. For detailed information on supplementary services, refer to
''Working with Supplementary Services'' on page 377.
¾ To configure the supplementary services' parameters, take these 4
steps:
1.
Open the 'Supplementary Services' page (Configuration tab > Protocol
Configuration menu > SIP Advanced Parameters submenu > Supplementary
Services page item).
Figure 3-65: Supplementary Services Page
2.
Configure the supplementary services parameters according to the table below.
3.
Click the Submit button to save your changes, or click the Subscribe to MWI or
Unsubscribe to MWI buttons to save your changes and to subscribe / unsubscribe to
the MWI server.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-36: Supplementary Services Parameters
Parameter
Enable Hold
[EnableHold]
Description
Enables interworking of the Hold/Retrieve supplementary service from PRI
to SIP.
ƒ
[0] Disable = Disables.
ƒ
[1] Enable = Enables (default).
Notes:
Enable Hold to ISDN
[EnableHold2ISDN]
ƒ
This capability is only supported by the Euro ISDN variant and only
from TE (user) to NT (network).
ƒ
To support interworking of the Hold/Retrieve supplementary service
from SIP to ISDN, set EnableHold2ISDN to 1.
Enables interworking of the Hold/Retrieve supplementary service from SIP
to PRI.
ƒ
[0] = Disabled (default)
ƒ
[1] = Enabled
Notes:
Version 5.6
ƒ
This capability is supported only for QSIG and Euro ISDN variants.
ƒ
To support interworking of the Hold/Retrieve supplementary service
from ISDN to SIP, set the parameter EnableHold to 1
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Parameter
Hold Format
[HoldFormat]
Description
Determines the format of the call hold request.
ƒ
[0] 0.0.0.0 = The connection IP address in SDP is 0.0.0.0 (default).
ƒ
[1] Send Only = The SDP contains the attribute 'a=sendonly'.
Note: This parameter is applicable only to QSIG and Euro ISDN
protocols.
Held Timeout
[HeldTimeout]
Enable Transfer
[EnableTransfer]
Determines the time interval that the device can allow a call to remain on
hold. If a Resume (un-hold Re-INVITE) message is received before the
timer expires, the call is renewed. If this timer expires, the call is released.
ƒ
[-1] = The call is placed on hold indefinitely until the initiator of on hold
retrieves the call again(default).
ƒ
[0 - 2400] =Time to wait in seconds, after which the call is released.
Determines whether call transfer is enabled.
ƒ
[0] Disable = Disable the call transfer service.
ƒ
[1] Enable = The device responds to a REFER message with the
Referred-To header to initiate a call transfer (default).
Notes:
Transfer Prefix
[xferPrefix]
ƒ
To use call transfer, the devices at both ends must support this option.
ƒ
To use call transfer, set the parameter EnableHold to 1.
Defines the string that is added as a prefix to the transferred / forwarded
called number when the REFER / 3xx message is received.
Notes:
Enable Call Forward
[EnableForward]
ƒ
The number manipulation rules apply to the user part of the REFERTO / Contact URI before it is sent in the INVITE message.
ƒ
This parameter can be used to apply different manipulation rules to
differentiate transferred number from the originally dialed number.
Determines whether Call Forward is enabled.
ƒ
[0] Disable = Disable the Call Forward service.
ƒ
[1] Enable = Enable Call Forward service(default).
The device doesn't initiate call forward, it can only respond to call forward
requests.
Note: To use this service, the devices at both ends must support this
option.
Enable Call Waiting
[EnableCallWaiting]
Determines whether Call Waiting is enabled.
ƒ
[0] Disable = Disable the Call Waiting service.
ƒ
[1] Enable = Enable the Call Waiting service (default).
If enabled, when the device initiates a Tel-to-IP call to a destination that is
busy, it plays a Call Waiting Ringback tone to the caller.
Notes:
SIP User's Manual
ƒ
The device's Call Progress Tones file must include a Call Waiting
Ringback tone.
ƒ
The EnableHold parameter must be enabled on the called side.
ƒ
For information on the Call Waiting feature, refer to Call Waiting.
ƒ
For information on the Call Progress Tones file, refer to Configuring the
Call Progress Tones File.
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Parameter
Description
Hook-Flash Code
[HookFlashCode]
Determines the digit pattern used by the PBX to indicate a Hook Flash
event. When this pattern is detected from the Tel side, the device
responds as if a Hook Flash event occurs and sends a SIP INFO
message if HookFlashOption is set to 1, indicating Hook Flash. If
configured and a Hook Flash indication is received from the IP side, the
device generates this pattern to the Tel side.
The valid range is a 25-character string. The default is a null string.
MLPP (Multilevel Precedence and Preemption) Note: For additional MLPP parameters, refer to
“Configuring the Digital Gateway Parameters” on page 207
Call Priority Mode
[CallPriorityMode]
Enables Priority Calls handling.
MLPP DiffServ
[MLPPDiffserv]
ƒ
[0] Disable = Disable (default).
ƒ
[1] MLPP = Priority Calls handling is enabled.
Defines the DiffServ value (differentiated services code point -DSCP) used in IP packets containing SIP messages that are
related to MLPP calls.
The valid range is 0 to 63. The default value is 50.
3.4.7.2.3 Stand-Alone Survivability
The 'SAS Configuration' page allows you to configure the device's Stand-Alone Survivability
(SAS) feature. This feature is useful for providing a local backup via the PSTN in Small or
Medium Enterprises (SME) that are serviced by IP Centrex services. In such environments,
the enterprise's incoming and outgoing telephone calls (external and internal) are controlled
by the Proxy, which communicates with the enterprise through the WAN interface. SAS
ensures that incoming, outgoing, and internal calls service is maintained in case of a WAN
or Proxy failure, using a PSTN (or an alternate VoIP) backup connection and the device's
built-in internal routing. To utilize the SAS feature, the VoIP CPEs such as IP phones or
residential gateways need to be defined so that their Proxy and Registrar destination
addresses and UDP port equal the SAS feature's IP address and SAS local SIP UDP port.
Notes:
Version 5.6
•
The 'SAS Configuration' page is Feature Key dependant and therefore is
available only if included in the device's Feature Key (refer to “Upgrading
the Software Upgrade Key” on page 233).
•
For a detailed explanation on SAS and for configuring various SAS
setups, refer to ''Stand-Alone Survivability (SAS) Feature'' on page 346).
•
For additional SAS parameters (configurable only using the ini file), refer
to ''SIP Configuration Parameters'' on page 284.
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¾ To configure the Stand-Alone Survivability parameters, take these
4 steps:
1.
Open the 'SAS Configuration' page (Configuration tab > Protocol Configuration
menu > SIP Advanced Parameters submenu > Stand-Alone Survivability page
item).
Figure 3-66: SAS Configuration Page
2.
Configure the parameters according to the table below.
3.
Click the Submit button to apply your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-37: Stand-Alone Survivability Parameters Description
Parameter
Enable SAS
[EnableSAS]
Description
Enables the Stand-Alone Survivability (SAS) feature.
ƒ
[0] Disable Disabled (default)
ƒ
[1] Enable = SAS is enabled
When enabled, the device receives the registration requests from
different SIP entities in the local network and then forwards them to the
defined proxy. If the connection to the proxy fails ('Emergency Mode'),
the device serves as a proxy by allowing calls internal to the local
network or outgoing to PSTN.
SAS Local SIP UDP Port
[SASLocalSIPUDPPort]
Local UDP port for sending and receiving SIP messages for SAS. The
SIP entities in the local network need to send the registration requests
to this port. When forwarding the requests to the proxy ('Normal
Mode'), this port serves as the source port.
The valid range is 1 to 65,534. The default value is 5080.
SAS Default Gateway IP
[SASDefaultGatewayIP]
The default gateway used in SAS 'Emergency Mode'. When an
incoming SIP INVITE is received and the destination Address-OfRecord is not included in the SAS database, the request is
immediately sent to this default gateway.
The address can be configured as an IP address (dotted-decimal
notation) or as a domain name (up to 49 characters). The default is a
null string, which is interpreted as the local IP address of the gateway.
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Parameter
Description
SAS Registration Time
[SASRegistrationTime]
Determines the value of the SIP Expires header that is sent in a 200
OK response to an incoming REGISTER message when in SAS
'Emergency Mode'.
The valid range is 10 to 2,000,000. The default value is 20.
Short Number Length
[SASShortNumberLength]
This parameter is obsolete; instead, use the parameter
SASRegistrationManipulation.
SAS Local SIP TCP Port
[SASLocalSIPTCPPort]
Local TCP port used to send/receive SIP messages for the SAS
application. The SIP entities in the local network need to send the
registration requests to this port. When forwarding the requests to the
proxy ('Normal Mode'), this port serves as the source port.
The valid range is 1 to 65,534. The default value is 5080.
SAS Local SIP TLS Port
[SASLocalSIPTLSPort]
Local TLS port used to send/receive SIP messages for the SAS
application. The SIP entities in the local network need to send the
registration requests to this port. When forwarding the requests to the
proxy ('Normal Mode'), this port serves as the source port.
The valid range is 1 to 65,534. The default value is 5081.
SAS Proxy Set
[SASProxySet]
Determines the Proxy Set (index number) used in SAS Normal mode
to forward REGISTER and INVITE requests from the users that are
served by the SAS application.
The valid range is 0 to 5. The default value is 0 (i.e., default Proxy
Set).
Redundant SAS Proxy Set
[RedundantSASProxySet]
Determines the Proxy Set (index number) used in SAS Emergency
mode for fallback when the user is not found in the Registered Users
database. Each time a new SIP request arrives, the SAS application
checks whether the user is listed in the registration database. If the
user is located in the database, the request is sent to the user. If the
user is not found, the request is forwarded to the next redundant SAS
defined in the Redundant SAS Proxy Set. If that SAS Proxy IP appears
in the Via header of the request, it is not forwarded (so that loops are
prevented in the request's course). If no such redundant SAS exists,
the SAS sends the request to its default gateway (configured by the
parameter SASDefaultGatewayIP).
The valid range is -1 to 5. The default value is -1 (i.e., no redundant
Proxy Set).
3.4.7.2.4 SBC Configuration
The 'SBC Settings' page allows you to enable the device's IP-to-IP call routing feature. To
enable IP-to-IP capabilities, the following prerequisites must be fulfilled:
„
The device must be loaded with the Feature Key that includes the SBC feature (refer
to ''Upgrading the Software Upgrade Key'' on page 233).
„
The device must be running SIP version 5.4 or later.
Version 5.6
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¾ To configure the SBC parameters, take these 4 steps:
1.
Open the 'SBC Settings' page (Configuration tab > Protocol Configuration menu >
SIP Advanced Parameters submenu > SBC Configuration page item).
Figure 3-67: SBC Settings Page
2.
Configure the SBC parameters according to the following table.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to Saving Configuration.
Table 3-38: SBC Parameters
Parameter
Description
Enable SBC
[EnableSBC]
Enables or disables the SBC feature.
SBC Registration Time
[SBCRegistrationTime]
ƒ
[0] Disable (default)
ƒ
[1] Enable
Configures the value (in sec) sent in the "expires" when the device replies
with a SIP 200 OK in response to Registration requests.
The default is 20.
Note: This parameter is applicable only to clients belonging to IP groups
of type "USER".
3.4.7.3
Configuring the Number Manipulation Tables
The device provides four Number Manipulation tables for incoming (IP-to-Tel) and outgoing
(Tel-to-IP) calls. These tables are used to modify the destination and source telephone
numbers so that the calls can be routed correctly. For example, telephone number
manipulation can be implemented for the following:
„
Strip or add dialing plan digits from or to the number. For example, a user may need to
first dial 9 before dialing the phone number to indicate an external line. This number
(9) can then be removed (by the Manipulation table) before the call is setup.
„
Allow or disallow Caller ID information to be sent according to destination or source
prefixes.
„
Assign NPI/TON to IP-to-Tel calls. The device can use a single global setting for
NPI/TON classification or it can use the setting in this table on a call-by-call basis.
The number manipulation is configured in the following tables:
„
For Tel-to-IP calls:
•
Destination Phone Number Manipulation Table for Tel-to-IP Calls
(NumberMapTel2IP ini file parameter)
•
Source Phone Number Manipulation Table for Tel-to-IP Calls
(SourceNumberMapTel2IP ini file parameter)
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For IP-to-Tel calls:
•
Destination Phone Number Manipulation Table for IP-to-Tel Calls
(NumberMapIP2Tel ini file parameter)
•
Source Phone Number Manipulation Table for IP-to-Tel Calls
(SourceNumberMapIP2Tel ini file parameter)
Notes:
•
Number manipulation can occur before or after a routing decision is
made. For example, you can route a call to a specific Trunk Group
according to its original number, and then you can remove or add a prefix
to that number before it is routed. To determine when number
manipulation is performed, configure the 'IP to Tel Routing Mode'
parameter (RouteModeIP2Tel) described in ''IP to Trunk Group Routing''
on page 181, and 'Tel to IP Routing Mode' parameter (RouteModeTel2IP)
described in ''Tel to IP Routing Table'' on page 175 (or “Outbound IP
Routing Table” on page 178).
•
For configuring number manipulation using ini file table parameters
NumberMapIP2Tel, NumberMapTel2IP, SourceNumberMapIP2Tel, and
SourceNumberMapTel2IP, refer to ''Number Manipulation and Routing
Parameters'' on page 313.
¾ To configure the Number Manipulation tables, take these 5 steps:
1.
Open the required 'Number Manipulation' page (Configuration tab > Protocol
Configuration menu > Manipulation Tables submenu > Dest Number IP->Tel, Dest
Number Tel->IP, Source Number IP->Tel, or Source Number Tel->IP page item);
the relevant Manipulation table page is displayed (e.g., 'Source Phone Number
Manipulation Table for TelÆIP Calls' page).
Figure 3-68: Source Phone Number Manipulation Table for Tel-to-IP Calls
The figure above shows an example of the use of manipulation rules in the 'Source
Phone Number Manipulation Table for TelÆIP Calls':
Version 5.6
•
When the destination number is 035000 and source number is 20155, the source
number is changed to 97120155.
•
When the source number is 1001876, it is changed to 587623.
•
When the source number is 1234510012001, it is changed to 20018.
•
When the source number is 3122, it is changed to 2312.
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2.
From the 'Table Index' drop-down list, select the range of entries that you want to edit
(up to 20 entries can be configured for Source Number IP-to-Tel Manipulation, up to
120 entries can be configured for Source Number Tel-to-IP Manipulation, and up to
100 entries for Destination Number Manipulation).
3.
Configure the Number Manipulation table according to the table below.
4.
Click the Submit button to save your changes.
5.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Notes:
•
The manipulation rules are executed in the following order:
1. Number of stripped digits.
2. Number of digits to leave.
3. Prefix / suffix to add.
•
The manipulation rules can be applied to any incoming call whose source
IP address (if applicable), source Trunk Group (if applicable), source IP
Group (if applicable), destination number prefix and source number prefix
matches the values defined in the 'Source IP Address', 'Source Trunk
Group', 'Source IP Group', 'Destination Prefix', and 'Source Prefix' fields
respectively. The number manipulation can be performed using a
combination of each of the above criteria, or using each criterion
independently.
•
For available notations that represent multiple numbers, refer to ''Dialing
Plan Notation'' on page 168.
Table 3-39: Number Manipulation Parameters Description
Parameter
Source Trunk Group
[_SrcTrunkGroupID]
Description
The source Trunk Group (1-99) for Tel-to-IP calls. To denote any
Trunk Group, leave this field empty.
Notes:
Source IP Group
[_SrcIPGroupID]
ƒ
This parameter is available only in the 'Source Phone Number
Manipulation Table for Tel -> IP Calls' and 'Destination Phone
Number Manipulation Table for Tel -> IP Calls' pages.
ƒ
For IP-to-IP call routing, this parameter is not required (i.e., leave
the field empty).
The IP Group from where the IP-to-IP call originated. Typically, this IP
Group of an incoming INVITE is determined/classified using the
‘Inbound IP Routing’ table. If not used (i.e., any IP Group), simply
leave the field empty.
Notes:
Destination Prefix
[_DestinationPrefix]
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ƒ
This parameter is available only in the 'Source Phone Number
Manipulation Table for Tel -> IP Calls' page.
ƒ
If this Source IP Group has a Serving IP Group, then all calls
originating from this Source IP Group is sent to the Serving IP
Group. In this scenario, this table is used only if the parameter
PreferRouteTable is set to 1.
Destination (called) telephone number prefix. An asterisk (*)
represents any number.
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Parameter
Description
Source Prefix
[_SourcePrefix]
Source (calling) telephone number prefix. An asterisk (*) represents
any number.
Source IP
[_SourceAddress]
Source IP address of the caller (obtained from the Contact header in
the INVITE message).
Notes:
ƒ
This parameter is applicable only to the Number Manipulation
tables for IP-to-Tel calls.
ƒ
The source IP address can include the 'x' wildcard to represent
single digits. For example: 10.8.8.xx represents all IP addresses
between 10.8.8.10 to 10.8.8.99.
ƒ
The source IP address can include the asterisk (*) wildcard to
represent any number between 0 and 255. For example, 10.8.8.*
represents all IP addresses between 10.8.8.0 and 10.8.8.255.
Stripped Digits From Left
[_RemoveFromLeft]
Number of digits to remove from the left of the telephone number
prefix. For example, if you enter 3 and the phone number is 5551234,
the new phone number is 1234.
Stripped Digits From Right
[_RemoveFromRight]
Number of digits to remove from the right of the telephone number
prefix. For example, if you enter 3 and the phone number is
5551234, the new phone number is 5551.
Prefix to Add
[_Prefix2Add]
The number or string that you want added to the front of the
telephone number. For example, if you enter '9' and the phone
number is 1234, the new number is 91234.
Suffix to Add
[_Suffix2Add]
The number or string that you want added to the end of the telephone
number. For example, if you enter '00' and the phone number is
1234, the new number is 123400.
Number of Digits to Leave
[_LeaveFromRight]
The number of digits that you want to retain from the right of the
phone number.
NPI
[_NumberPlan]
The Numbering Plan Indicator (NPI) assigned to this entry.
ƒ
[0] Unknown (default)
ƒ
[9] Private
ƒ
[1] E.164 Public
ƒ
[-1] Not Configured = value received from PSTN/IP is used
Notes:
TON
[_NumberType]
Version 5.6
ƒ
This parameter is applicable only to Number Manipulation tables
for IP-to-Tel calls.
ƒ
For a detailed list of the available NPI/TON values, refer to
Numbering Plans and Type of Number on page 169
The Type of Number (TON) assigned to this entry.
ƒ
If you selected 'Unknown' for the NPI, you can select Unknown
[0].
ƒ
If you selected 'Private' for the NPI, you can select Unknown [0],
Level 2 Regional [1], Level 1 Regional [2], PISN Specific [3] or
Level 0 Regional (Local) [4].
ƒ
If you selected 'E.164 Public' for the NPI, you can select Unknown
[0], International [1], National [2], Network Specific [3], Subscriber
[4] or Abbreviated [6].
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Parameter
Description
Notes:
Presentation
[_IsPresentationRestricted]
ƒ
This parameter is applicable only to Number Manipulation tables
for IP-to-Tel calls.
ƒ
The default is 'Unknown'.
Determines whether Caller ID is permitted:
ƒ
Not Configured = privacy is determined according to the Caller ID
table (refer to Caller ID).
ƒ
Allowed = sends Caller ID information when a call is made using
these destination / source prefixes.
ƒ
Restricted = restricts Caller ID information for these prefixes.
Notes:
ƒ
Only applicable to Number Manipulation tables for source number
manipulation.
ƒ
If 'Presentation' is set to 'Restricted' and 'Asserted Identity Mode'
is set to 'P-Asserted', the From header in the INVITE message
includes the following: From: 'anonymous' <sip:
[email protected]> and 'privacy: id' header.
3.4.7.3.1 Dialing Plan Notation
The dialing plan notation applies to the Number Manipulation tables, 'Tel to IP Routing'
table (refer to ''Tel to IP Routing Table'' on page 175), and 'IP to Trunk Group Routing' table
(refer to ''IP to Trunk Group Routing'' on page 181). The dialing notation applies to digits
entered for the destination and source prefixes to represent multiple numbers.
Table 3-40: Dialing Plan Notations
Notation
[n-m]
[n,m,...]
Description
Example
Represents a range of
numbers.
Note: Range of letters is
not supported.
ƒ
[5551200-5551300]#: represents all numbers from
5551200 to 5551300.
ƒ
123[100-200]#: represents all numbers from
123100 to 123200.
Represents multiple
numbers. Up to three digits
can be used to denote
each number.
ƒ
[2,3,4,5,6]#: represents a one-digit number that
starts with 2, 3, 4, 5, or 6.
ƒ
[11,22,33]xxx#: represents a four-digit number
that starts 11, 22, or 33.
ƒ
[111,222]xxx#: represents a four-digit number that
starts 111 or 222.
x
Represents any single
digit.
54324: represents any number that starts with 54324.
Pound sign
(#)
at the end of
a number
Represents the end of a
number.
54324xx#: represents a 7-digit number that starts with
54324.
A single
asterisk (*)
Represents any number.
*: represents any number (i.e., all numbers).
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The device matches the rules starting at the top of the table (i.e., top rules take precedence
over lower rules). For this reason, enter more specific rules above more generic rules. For
example, if you enter 551 in entry 1 and 55 in entry 2, the device applies rule 1 to numbers
that start with 551 and applies rule 2 to numbers that start with 550, 552, 553, 554, 555,
556, 557, 558 and 559. However, if you enter 55 in entry 1 and 551 in entry 2, the device
applies rule 1 to all numbers that start with 55 including numbers that start with 551.
3.4.7.3.2 Numbering Plans and Type of Number
Numbers are classified by their Numbering Plan Indication (NPI) and their Type of Number
(TON). The device supports all NPI/TON classifications used in the standard. The list of
ISDN ETSI NPI/TON values is shown in the following table:
Table 3-41: NPI/TON Values for ISDN ETSI
NPI
TON
Description
Unknown [0]
Unknown [0]
A valid classification, but one that has no information
about the numbering plan.
E.164 Public
[1]
Unknown [0]
A public number in E.164 format, but no information
on what kind of E.164 number.
International [1]
National [2]
Subscriber [4]
Private [9]
Unknown [0]
A public number in complete international E.164
format, e.g., 16135551234.
A public number in complete national E.164 format,
e.g., 6135551234.
A public number in complete E.164 format
representing a local subscriber, e.g., 5551234.
A private number, but with no further information
about the numbering plan.
Level 2 Regional [1]
Level 1 Regional [2]
A private number with a location, e.g., 3932200.
PISN Specific [3]
Level 0 Regional (local) [4]
A private local extension number, e.g., 2200.
For NI-2 and DMS-100 ISDN variants, the valid combinations of TON and NPI for calling
and called numbers include (Plan/Type):
„
0/0 - Unknown/Unknown
„
1/1 - International number in ISDN/Telephony numbering plan
„
1/2 - National number in ISDN/Telephony numbering plan
„
1/4 - Subscriber (local) number in ISDN/Telephony numbering plan
„
9/4 - Subscriber (local) number in Private numbering plan
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3.4.7.3.3 Mapping NPI/TON to Phone-Context
The 'Phone-Context Table' page is used to map NPI and TON to the Phone-Context SIP
parameter. When a call is received from the ISDN, the NPI and TON are compared against
the table and the Phone-Context value is used in the outgoing SIP INVITE message. The
same mapping occurs when an INVITE with a Phone-Context attribute is received. The
Phone-Context parameter appears in the standard SIP headers where a phone number is
used (Request-URI, To, From, Diversion).
¾ To configure the Phone-Context tables, take these 4 steps:
1.
Open the 'Phone Context Table' page (Configuration tab > Protocol Configuration
menu > Manipulation Tables submenu > Phone Context Table page item).
Figure 3-69: Phone Context Table Page
2.
Configure the Phone Context table according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Notes:
•
Several rows with the same NPI-TON or Phone-Context are allowed. In
such a scenario, a Tel-to-IP call uses the first match.
•
Phone-Context '+' is a unique case as it doesn't appear in the RequestURI as a Phone-Context parameter. Instead, it's added as a prefix to the
phone number. The '+' isn't removed from the phone number in the IP-toTel direction.
•
You can also configure the Phone Context table using the ini file table
parameter PhoneContext (refer to ''Number Manipulation and Routing
Parameters'' on page 313).
Table 3-42: Phone-Context Parameters Description
Parameter
Description
Add Phone Context As Prefix
[AddPhoneContextAsPrefix]
Determines whether the received Phone-Context parameter is
added as a prefix to the outgoing ISDN SETUP message with Called
and Calling numbers.
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ƒ
[0] Disable = Disable (default).
ƒ
[1] Enable = Enable.
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Parameter
Description
Select the Number Plan assigned to this entry.
NPI
ƒ
[0] Unknown = Unknown (default)
ƒ
[1] E.164 Public = E.164 Public
ƒ
[9] Private = Private
For a detailed list of the available NPI/TON values, refer to
Numbering Plans and Type of Number on page 169.
Select the Number Type assigned to this entry.
TON
Phone Context
3.4.7.4
ƒ
If you selected Unknown as the NPI, you can select Unknown [0].
ƒ
If you selected Private as the NPI, you can select Unknown [0],
Level 2 Regional [1], Level 1 Regional [2], PSTN Specific [3], or
Level 0 Regional (Local) [4].
ƒ
If you selected E.164 Public as the NPI, you can select Unknown
[0], International [1], National [2], Network Specific [3],
Subscriber [4], or Abbreviated [6].
The Phone-Context SIP URI parameter.
Configuring the Routing Tables
The Routing Tables submenu allows you to configure the device's call routing. This
submenu includes the following page items:
„
Routing General Parameters (refer to ''Routing General Parameters'' on page 171)
„
Tel to IP Routing (refer to ''Tel to IP Routing Table'' on page 175)
„
Outbound IP Routing (refer to “Outbound IP Routing Table” on page 178)
„
IP to Trunk Group Routing (refer to ''IP to Trunk Group Routing'' on page 181)
„
Inbound IP Routing (refer to “Inbound IP Routing Table” on page 184)
„
Internal DNS Table (refer to ''Internal DNS Table'' on page 186)
„
Internal SRV Table (refer to ''Internal SRV Table'' on page 187)
„
Reasons for Alternative Routing (refer to ''Reasons for Alternative Routing'' on page
188)
„
Release Cause Mapping (refer to “Release Cause Mapping” on page 189)
3.4.7.4.1 Routing General Parameters
The 'Routing General Parameters' page allows you to configure the device's IP-to-Tel and
Tel-to-IP routing parameters.
¾ To configure the general routing parameters, take these 4 steps:
1.
Version 5.6
Open the 'Routing General Parameters' page (Configuration tab > Protocol
Configuration menu > Routing Tables submenu > Routing General Parameters
page item).
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Figure 3-70: Routing General Parameters Page
2.
Configure the general parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-43: Routing General Parameters Description
Parameter
Add Trunk Group ID as Prefix
[AddTrunkGroupAsPrefix]
Description
Determines whether the device's Trunk Group ID is added
as a prefix to the destination phone number for Tel-to-IP
calls.
ƒ
[0] No = Don't add Trunk Group ID as prefix (default).
ƒ
[1] Yes = Add Trunk Group ID as prefix to called number.
Notes:
Add Trunk ID as Prefix
[AddPortAsPrefix]
ƒ
This option can be used to define various routing rules.
ƒ
To use this feature, you must configure the Trunk Group
IDs (refer to “Configuring the Trunk Group Table” on
page 195).
Determines whether the Trunk ID is added as a prefix to the
called number for Tel-to-IP calls.
ƒ
[0] No = Don't add Trunk ID as prefix (default).
ƒ
[1] Yes = Enable add Trunk ID as prefix.
If enabled, the Trunk ID (single digit in the range 1 to 8 ) is
added as a prefix to the called (destination) phone number.
This option can be used to define various routing rules.
Replace Empty Destination with Bchannel Phone Number
[ReplaceEmptyDstWithPortNumber]
Determines whether the internal channel number is used as
the destination number if the called number is missing.
ƒ
[0] No (default)
ƒ
[1] Yes
Note: Applicable only for Tel-to-IP calls and if the called
number is missing.
Add NPI and TON to Calling Number
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Determines whether Numbering Plan Indicator (NPI) and
Type of Numbering (TON) are added to the Calling Number
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Parameter
[AddNPIandTON2CallingNumber]
Description
for Tel-to-IP calls.
ƒ
[0] No = Do not change the Calling Number (default).
ƒ
[1] Yes = Add NPI and TON to the Calling Number ISDN
Tel-to-IP call.
For example: After receiving a Calling Number of 555, NPI
of 1, and TON of 3, the modified number becomes 13555.
This number can later be used for manipulation and routing.
Add NPI and TON to Called Number
[AddNPIandTON2CalledNumber]
Determines whether NPI and TON are added to the Called
Number for Tel-to-IP calls.
ƒ
[0] No = Do not change the Called Number (default).
ƒ
[1] Yes = Add NPI and TON to the Called Number of
ISDN Tel-to-IP call.
For example: After receiving a Called Number of 555, NPI of
1 and TON of 3, the modified number becomes 13555. This
number can later be used for manipulation and routing.
IP to Tel Remove Routing Table Prefix
[RemovePrefix]
Determines whether the device removes the prefix from the
destination number for IP-to-Tel calls.
ƒ
[0] No = Don't remove prefix (default)
ƒ
[1] Yes = Remove the prefix (defined in the 'IP to Trunk
Group Routing' table - refer to ''IP to Trunk Group
Routing'' on page 181) from a telephone number for an
IP-to-Tel call, before forwarding it to Tel.
For example: To route an incoming IP-to-Tel call with
destination number 21100, the 'IP to Trunk Group Routing'
table is scanned for a matching prefix. If such a prefix is
found (e.g., 21), then before the call is routed to the
corresponding Trunk Group, the prefix (21) is removed from
the original number, and therefore, only 100 remains.
Notes:
Source IP Address Input
[SourceIPAddressInput]
ƒ
Applicable only if number manipulation is performed after
call routing for IP-to-Tel calls (i.e., RouteModeIP2Tel
parameter is set to 0).
ƒ
Similar operation (of removing the prefix) is also
achieved by using the usual number manipulation rules.
Determines the IP address that the device uses to
determine the source of incoming INVITE messages for IPto-Tel routing.
ƒ
[-1] = Not configured (default).
ƒ
[0] SIP Contact Header = Use the IP address received in
the Contact header of the incoming INVITE message.
ƒ
[1] Layer 3 Source IP = Use the actual IP address (Layer
3) from which the SIP packet was received.
Note: If the IP-to-IP feature is enabled (i.e., supported by
the Feature Key and EnableSBC is set to 1 - refer to “SBC
Configuration” on page 163), this parameter is automatically
set to 1. If the IP-to-IP feature is disabled, this parameter is
automatically set to 0.
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Parameter
Enable Alt Routing Tel to IP
[AltRoutingTel2IPEnable]
Description
Enables the Alternative Routing feature for Tel-to-IP calls.
ƒ
[0] Disable = Disables the Alternative Routing feature
(default).
ƒ
[1] Enable = Enables the Alternative Routing feature.
ƒ
[2] Status Only = The Alternative Routing feature is
disabled, but read-only information on the Quality of
Service of the destination IP addresses is provided.
For information on the Alternative Routing feature, refer to
''Configuring Alternative Routing (Based on Connectivity and
QoS)'' on page 361.
Alt Routing Tel to IP Mode
[AltRoutingTel2IPMode]
Determines the event(s) reason for triggering Alternative
Routing.
ƒ
[0] None = Alternative routing is not used.
ƒ
[1] Connectivity = Alternative routing is performed if ping
to initial destination fails.
ƒ
[2] QoS = Alternative routing is performed if poor QoS is
detected.
ƒ
[3] Both = Alternative routing is performed if either ping
to initial destination fails, poor Quality of Service is
detected, or DNS host name is not resolved (default).
Notes:
Alt Routing Tel to IP Connectivity
Method
[AltRoutingTel2IPConnMethod]
ƒ
QoS is quantified according to delay and packet loss
calculated according to previous calls. QoS statistics are
reset if no new data is received within two minutes. For
information on the Alternative Routing feature, refer to
''Configuring Alternative Routing (Based on Connectivity
and QoS)'' on page 361.
ƒ
To receive quality information (displayed in the 'Quality
Status' and 'Quality Info.' fields in ''IP Connectivity'' on
page 252) per destination, this parameter must be set to
2 or 3.
Determines the method used by the device for periodically
querying the connectivity status of a destination IP address.
ƒ
[0] ICMP Ping (default) = Internet Control Message
Protocol (ICMP) ping messages.
ƒ
[1] SIP OPTIONS = The remote destination is
considered offline if the latest OPTIONS transaction
timed out. Any response to an OPTIONS request, even if
indicating an error, brings the connectivity status to
online.
Alt Routing Tel to IP Keep Alive Time
[AltRoutingTel2IPKeepAliveTime]
Defines the time interval (in seconds) between SIP
OPTIONS Keep-Alive messages used for the IP
Connectivity application.
The valid range is 5 to 2,000,000. The default value is 60.
Max Allowed Packet Loss for Alt
Routing [%]
[IPConnQoSMaxAllowedPL]
Packet loss percentage at which the IP connection is
considered a failure and Alternative Routing mechanism is
activated.
The range is 1 to 20%. The default value is 20%.
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Parameter
Max Allowed Delay for Alt Routing
[msec]
[IPConnQoSMaxAllowedDelay]
Description
Transmission delay (in msec) at which the IP connection is
considered a failure and Alternative Routing mechanism is
activated.
The range is 100 to 1000. The default value is 250.
3.4.7.4.2 Tel to IP Routing Table
The 'Tel to IP Routing' page provides a table for configuring up to up to 50 routing rules for
Tel-to-IP calls, where Tel calls are routed to destinations based on IP address (or IP
Group).
Note: The 'Tel to IP Routing' page appears only if the parameter EnableSBC is set
to 0 (default) in ''SBC Configuration'' on page 163. If this parameter is
enabled, the 'Outbound IP Routing Table' page appears instead (refer to
''Outbound IP Routing Table'' on page 178 for a description of this page).
This routing table associates called and/or calling telephone number prefixes (originating
from a specific Trunk Group), with a destination IP address (or Fully Qualified Domain
Name - FQDN) or IP Group. When a call is routed by the device (i.e., a Proxy server isn't
used), the called and calling numbers are compared to the list of prefixes in this table. Calls
that match these prefixes are sent to the corresponding IP address. If the number dialed
does not match these prefixes, the call is not made.
When using a Proxy server, you do not need to configure this table unless you require one
of the following:
„
Fallback routing when communication with Proxy servers is lost.
„
Implement the 'Filter Calls to IP' and 'IP Security' features.
„
Obtain different SIP URI host names (per called number).
„
Assign IP profiles.
Note that for this table to take precedence over a Proxy for routing calls, set the parameter
PreferRouteTable to 1. The device checks the 'Destination IP Address' field in this table for
a match with the outgoing call. A Proxy is used only if a match is not found.
Possible uses for Tel-to-IP routing include the following:
„
Fallback to internal routing table if there is no communication with the Proxy servers.
„
Call Restriction (when Proxy isn't used): rejects all outgoing Tel-to-IP calls that are
associated with the destination IP address 0.0.0.0.
„
IP Security: When the IP Security feature is enabled (SecureCallFromIP = 1), the
device accepts only those IP-to-Tel calls with a source IP address defined in the 'Tel to
IP Routing' table.
„
Filter Calls to IP: When a Proxy is used, the device checks the 'Tel to IP Routing' table
before a telephone number is routed to the Proxy. If the number is not allowed
(number isn't listed or a Call Restriction routing rule is applied), the call is released.
„
Always Use Routing Table: When this feature is enabled (AlwaysUseRouteTable = 1),
even if a Proxy server is used, the SIP URI host name in the sent INVITE message is
obtained from this table. Using this feature, you can assign a different SIP URI host
name for different called and/or calling numbers.
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„
Assign Profiles to destination addresses (also when a Proxy is used).
„
Alternative Routing (when a Proxy isn't used): an alternative IP destination for
telephone number prefixes is available. To associate an alternative IP address to a
called telephone number prefix, assign it with an additional entry (with a different IP
address), or use an FQDN that resolves into two IP addresses. The call is sent to the
alternative destination when one of the following occurs:
•
No ping to the initial destination is available, poor QoS (delay or packet loss,
calculated according to previous calls) is detected, or a DNS host name is not
resolved. For detailed information on Alternative Routing, refer to ''Configuring
Alternative Routing (Based on Connectivity and QoS'' on page 361.
•
A release reason defined in the 'Reasons for Alternative Tel to IP Routing' table is
received (refer to ''Reasons for Alternative Routing'' on page 188).
Alternative routing (using this table) is commonly implemented when there is no
response to an INVITE message (after INVITE retransmissions). The device then
issues an internal 408 'No Response' implicit release reason. If this reason is included
in the 'Reasons for Alternative Routing' table, the device immediately initiates a call to
the redundant destination using the next matched entry in the 'Tel to IP Routing' table.
Note that if a domain name in this table is resolved into two IP addresses, the timeout
for INVITE retransmissions can be reduced by using the parameter 'Number of RTX
Before Hotswap'.
Notes:
•
If the alternative routing destination is the device itself, the call can be
configured to be routed back to the PSTN. This feature is referred to as
'PSTN Fallback', meaning that if poor voice quality occurs over the IP
network, the call is routed through the legacy telephony system (PSTN).
•
Tel-to-IP routing can be performed before or after applying the number
manipulation rules. To control when number manipulation is performed,
use the 'Tel to IP Routing Mode' (or RouteModeTel2IP ini file) parameter,
described in the table below.
•
You can also configure the 'Tel to IP Routing' table using the ini file table
parameter Prefix (refer to ''Number Manipulation and Routing
Parameters'' on page 313).
¾ To configure the Tel to IP Routing table, take these 5 steps:
1.
Open the 'Tel to IP Routing' page (Configuration tab > Protocol Configuration menu
> Routing Tables submenu > Tel to IP Routing page item).
Figure 3-71: Tel to IP Routing Page
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2.
From the 'Routing Index' drop-down list, select the range of entries that you want to
add.
3.
Configure the Tel to IP Routing table according to the table below.
4.
Click the Submit button to save your changes.
5.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-44: Tel to IP Routing Table Parameters Description
Parameter
Tel to IP Routing Mode
[RouteModeTel2IP]
Description
Determines whether to route Tel calls to IP before or after
manipulation of destination number.
ƒ
[0] Route calls before manipulation = Tel-to-IP calls are routed
before the number manipulation rules are applied (default).
ƒ
[1] Route calls after manipulation = Tel-to-IP calls are routed after
the number manipulation rules are applied.
Notes: Not applicable if outbound Proxy routing is used.
Src. Trunk Group ID
[PREFIX_SrcTrunkGroupID]
The source Trunk Group for Tel-to-IP calls.
The range is 1-99.
Notes:
ƒ
If this parameter is not required in the routing rule, leave the field
empty.
ƒ
To denote any Trunk Group, you can enter the asterisk (*)
symbol.
Dest. Phone Prefix
[PREFIX_DestinationPrefix]
Represents a called telephone number prefix. The prefix can be 1 to
19 digits long. An asterisk (*) represents all numbers.
Source Phone Prefix
[PREFIX_SourcePrefix]
Represents a calling telephone number prefix. The prefix can be 1 to
19 digits long. An asterisk (*) represents all numbers.
All Tel calls matching all or any combination of the above routing rules are subsequently sent to the
destination IP address defined below.
Notes:
ƒ
For alternative routing, additional entries of the same prefixes can be configured.
ƒ
For notations representing multiple numbers, refer to ''Dialing Plan Notation'' on page 168.
Dest. IP Address
[PREFIX_DestAddress]
The destination IP address (in dotted decimal notation) to where
these calls must be sent. Domain names (e.g., domain.com) can be
used instead of IP addresses.
Notes:
Version 5.6
ƒ
If you select a destination IP Group (in the 'Dest IP Group ID' field
below), then the IP address you define in this 'Dest IP Address'
field is not used for routing and therefore, not required.
ƒ
To discard outgoing IP calls of a specific Tel-to-IP routing rule,
enter 0.0.0.0. For example, if you want to prohibit dialing of
international calls, then in the 'Dest Phone Prefix' field, enter 00
and in the 'Dest IP Address' field, enter 0.0.0.0.
ƒ
For routing calls between phones connected to the device (i.e.,
local routing), enter the device's IP address. When the device's IP
address is unknown (e.g., when DHCP is used), enter the IP
address 127.0.0.1.
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Parameter
Description
ƒ
When using domain names, you must enter a DNS server IP
address or alternatively, define these names in the 'Internal DNS
Table' (refer to ''Internal DNS Table'' on page 186).
Port
[PREFIX_DestPort]
The destination port to where you want to route the Tel-to-IP call.
Transport Type
[PREFIX_TransportType]
The transport layer type for sending the Tel-to-IP calls:
ƒ
[-1] Not Configured
ƒ
[0] UDP
ƒ
[1] TCP
ƒ
[2] TLS
Note: When 'Not Configured' is selected, the transport type defined
by the parameter SIPTransportType (refer to ''SIP General
Parameters'' on page 121) is used.
Dest IP Group ID
[PREFIX_DestIPGroupID]
The IP Group (1-9) to where you want to route the Tel-to-IP call. The
SIP INVITE messages are sent to the IP address(es) of the Proxy
Set that is associated with the selected IP Group.
If you select an IP Group, it is unnecessary to configure a destination
IP address (in the 'Dest IP Address' field). However, if both
parameters are configured, the INVITE message is sent only to the
IP Group.
If the parameter AlwaysUseRouteTable is set to 1 (in the 'IP Group'
table, refer to ''Configuring the IP Groups'' on page 201), the request
URI host name in the INVITE message is set to the value of the
parameter 'Dest IP Address' (if not empty); otherwise, it is set to the
value of the parameter 'SIP Group Name' (defined in the 'IP Group'
table).
Note: To configure Proxy Sets, refer to ''Proxy Sets Table'' on page
141.
IP Profile ID
[PREFIX_ProfileId]
The IP Profile ID (configured in ''Configuring the Profile Definitions''
on page 190) assigned to this routing rule entry for the IP destination.
Status
A read-only field representing the Quality of Service of the
destination IP address:
ƒ
n/a = Alternative Routing feature is disabled.
ƒ
OK = IP route is available.
ƒ
Ping Error = No ping to IP destination; route is not available.
ƒ
QoS Low = Bad QoS of IP destination; route is not available.
ƒ
DNS Error = No DNS resolution (only when domain name is used
instead of an IP address).
3.4.7.4.3 Outbound IP Routing Table
The 'Outbound IP Routing Table' page allows you to configure the device for routing
outbound (i.e., sent) IP-to-IP calls. This table routes inbound IP calls (identified in ''Inbound
IP Routing Table'' on page 184) received from an IP Group (refer to ''Configuring the IP
Groups'' on page 201) to a specific IP Group destination (or IP address).
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Note: The 'Outbound IP Routing Table' page appears only if the parameter
EnableSBC is set to 1 (i.e., enabled) in ''SBC Configuration'' on page 163. If
this parameter is not enabled (default), the 'Tel to IP Routing' page appears
instead (refer to ''Tel to IP Routing Table'' on page 175 for a description of this
page).
This table allows you to configure the device's routing rules for sending inbound IP calls
matching some or all of the following criteria to a destination IP address or IP Group:
„
Source IP Group
„
Source host prefix
„
Destination host prefix
„
Trunk Group
„
Destination telephone prefix
„
Source telephone prefix
¾ To configure Outbound IP Routing, take these 5 steps:
1.
Open the 'Outbound IP Routing Table' page (Configuration tab > Protocol
Configuration menu > Routing Tables submenu > Tel to IP Routing page item).
Figure 3-72: Outbound IP Routing Page
2.
From the 'Routing Index' drop-down list, select the range of entries that you want to
add.
3.
Configure the Outbound IP Routing table according to the table below.
4.
Click the Submit button to save your changes.
5.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-45: Outbound IP Routing Table Description
Parameter
Tel to IP Routing Mode
[RouteModeTel2IP]
Description
Determines whether to route the inbound IP calls to the IP
destination before or after manipulation of destination number.
ƒ
[0] Route calls before manipulation = IP-to-IP calls are routed
before the number manipulation rules are applied (default).
ƒ
[1] Route calls after manipulation = IP-to-IP calls are routed after
the number manipulation rules are applied.
Note: Not applicable if outbound Proxy routing is used.
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Parameter
Src. IPGroupID
[PREFIX_SrcIPGroupID]
Description
The IP Group ID from where the IP-to-IP call originated. Typically,
the IP Group of an incoming INVITE is determined according to the
‘Inbound IP Routing’ table. To denote all IP Groups, leave the field
empty.
Notes:
ƒ
If this Source IP Group has a Serving IP Group, then all calls
originating from this Source IP Group are sent to the Serving IP
Group. In this scenario, this table is used only if the parameter
PreferRouteTable is set to 1.
ƒ
For defining IP Groups, refer to ''Configuring the IP Groups'' on
page 201.
Src. Host Prefix
[PREFIX_SrcHostPrefix]
The prefix of the SIP URI host name in the From header of the
incoming SIP INVITE message. If this routing rule is not required,
leave the field empty. To denote any prefix, use the asterisk (*)
symbol
Dest. Host Prefix
[PREFIX_DestHostPrefix]
The request SIP URI host name prefix of the incoming SIP INVITE
message. If this routing rule is not required, leave the field empty. To
denote any prefix, use the asterisk (*) symbol
Src. Trunk Group ID
[PREFIX_SrcTrunkGroupID]
The source Trunk Group (1-99) for Tel-to-IP calls. For IP-to-IP calls,
this parameter is not required (i.e., leave the field empty). To denote
any Trunk Group, leave this field empty.
Note: For defining Trunk Groups, refer to ''Configuring the Trunk
Group Table'' on page 195.
Dest. Phone Prefix
[PREFIX_DestinationPrefix]
Represents a called telephone number prefix. The prefix can be 1 to
19 digits long. An asterisk (*) represents all numbers.
Source Phone Prefix
[PREFIX_SourcePrefix]
Represents a calling telephone number prefix. The prefix can be 1 to
19 digits long. An asterisk (*) represents all numbers.
Outbound IP calls matching all or any combination of the above routing rules are subsequently sent to
the destination IP address or IP Group defined below.
Notes:
ƒ
For alternative routing, additional entries of the same prefixes can be configured.
ƒ
For notations representing multiple numbers, refer to ''Dialing Plan Notation'' on page 168.
Dest. IP Address
[PREFIX_DestAddress]
Port
[PREFIX_DestPort]
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The destination IP address to where the outbound call is sent.
Domain names (e.g., domain.com) can be used instead of IP
addresses.
Notes:
ƒ
If you select a destination IP Group (in the 'Dest IP Group ID' field
below), then the IP address you define in this 'Dest IP Address'
field is not used for routing and therefore, not required.
ƒ
When using domain names, you must enter a DNS server IP
address or alternatively, define these names in the 'Internal DNS
Table' (refer to ''Internal DNS Table'' on page 186).
ƒ
To discard outgoing IP calls, define the IP address as 0.0.0.0.
ƒ
The IP address 127.0.0.1 can be used when the IP address of the
device itself is unknown (for example, when DHCP is used).
The destination port.
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Parameter
Transport Type
[PREFIX_TransportType]
Description
The transport layer type for sending the outbound SIP IP calls:
ƒ
[-1] Not Configured
ƒ
[0] UDP
ƒ
[1] TCP
ƒ
[2] TLS
Note: When 'Not Configured' is selected, the transport type defined
by the parameter SIPTransportType (refer to ''SIP General
Parameters'' on page 121) is used.
Dest. IP Group ID
[PREFIX_DestIPGroupID]
The IP Group (1 to 9) to where you want to route the outbound IP-toIP call. The INVITE messages are sent to the IP address(es) defined
for the Proxy Set that is associated with this IP Group. If you select
an IP Group, it is unnecessary to configure a destination IP address
(in the 'Dest IP Address' field above). However, if both parameters
are configured, the INVITE message is sent only to the IP Group.
If the destination IP Group is of type USER, the device searches for
a match between the request URI (of the received INVITE) to an
AOR registration record in the device's internal database. The
INVITE is then sent to the IP address of the registered contact.
If the parameter 'AlwaysUseRouteTable' (AlwaysUseRouteTable) is
set to 'Enable' (1) in the ‘IP Group’ table (refer to ''Configuring the IP
Groups'' on page 201), the request SIP URI host name in the INVITE
message is set to the value of the parameter 'Dest IP Address' (if
defined); otherwise, it is set to the value of the parameter 'SIP Group
Name' (defined in the ‘IP Group’ table).
Note: This parameter is also used as the ‘Serving IP Group’ in the
‘Account’ table for acquiring authentication user/password for this
call.
IP Profile ID
[PREFIX_ProfileId]
IP Profile ID (defined in ''IP Profile Settings'' on page 193) assigned
to the outbound IP call. This allows you to assign many different
configuration attributes (e.g., voice coders) to this IP Group outbound
routing rule.
Status
A read-only field representing the Quality of Service of the
destination IP address:
ƒ
n/a = Alternative Routing feature is disabled.
ƒ
OK = IP route is available.
ƒ
Ping Error = No ping to IP destination; route is not available.
ƒ
QoS Low = Bad QoS of IP destination; route is not available.
ƒ
DNS Error = No DNS resolution (only when domain name is used
instead of an IP address).
3.4.7.4.4 IP to Trunk Group Routing Table
The 'IP to Trunk Group Routing Table' page provides a table for routing incoming IP calls
to groups of channels (E1/T1 B-channels)called Trunk Groups. Trunk Group ID's are
assigned to the device's channels in the 'Trunk Group Table' page (refer to “Configuring the
Trunk Group Table” on page 195). You can add up to 24 IP-to-Trunk Group routing rules in
the table.
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Note: The 'IP to Trunk Group Routing Table' page appears only if the parameter
EnableSBC is set to 0 (default) in ''SBC Configuration'' on page 163. If this
parameter is enabled, the 'Inbound IP Routing Table' page appears instead
(refer to ''Inbound IP Routing Table'' on page 184 for a description of this
page).
The IP-to-Tel calls are routed to Trunk Groups according to any one of the following (or a
combination thereof) criteria:
„
Destination and source host prefix
„
Destination and source phone prefix
„
Source IP address
Once the call is routed to the specific Trunk Group, the call is sent to the device's channels
pertaining to that Trunk Group. The specific channel within the Trunk Group to which the
call is sent is determined according to the Trunk Group's channel selection mode. This
channel selection mode can be defined per Trunk Group (refer to ''Configuring the Trunk
Group Settings'' on page 197) or for all Trunk Groups using the global parameter
ChannelSelectMode.(refer to ''SIP General Parameters'' on page 121).
Notes:
•
When a call release reason (defined in ''Reasons for Alternative Routing''
on page 188) is received for a specific IP-to-Tel call, an alternative Trunk
Group for that call can be configured. This is performed by assigning the
call to an additional routing rule in the table (i.e., repeat the same routing
rule, but with a different Trunk Group ID).
•
You can also configure the 'IP to Trunk Group Routing' table using the ini
file table parameter PSTNPrefix (refer to ''Number Manipulation and
Routing Parameters'' on page 313).
¾ To configure the IP to Trunk Group Routing table, take these 5
steps:
1.
Open the 'IP to Trunk Group Routing' page (Configuration tab > Protocol
Configuration menu > Routing Tables submenu > IP to Trunk Group Routing page
item).
Figure 3-73: IP to Trunk Group Routing Table Page
2.
From the 'Routing Index' drop-down list, select the range of entries that you want to
add.
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3.
Configure the table according to the table below.
4.
Click the Submit button to save your changes.
5.
To save the changes so they are available after a power failure, refer to ''Saving
Configuration'' on page 230.
Table 3-46: IP to Trunk Group Routing Table Description
Parameter
IP to Tel Routing Mode
[RouteModeIP2Tel]
Dest. Host Prefix
[PstnPrefix_DestHostPrefix]
Description
Determines whether to route IP calls to the Trunk Group before or
after manipulation of destination number (configured in ''Configuring
the Number Manipulation Tables'' on page 164).
ƒ
[0] Route calls before manipulation = IP-to-Tel calls are routed
before the number manipulation rules are applied (default).
ƒ
[1] Route calls after manipulation = IP-to-Tel calls are routed
after the number manipulation rules are applied.
The request URI host name prefix of the incoming SIP INVITE
message. If this routing rule is not required, leave the field empty.
Note: For notations representing multiple numbers, refer to ''Dialing
Plan Notation'' on page 168. However, the asterisk (*) wildcard
cannot be used to depict any source host prefix.
Source Host Prefix
[PstnPrefix_SrcHostPrefix]
The From URI host name prefix of the incoming SIP INVITE
message. If this routing rule is not required, leave the field empty..
Notes:
Dest. Phone Prefix
[PstnPrefix_DestPrefix]
ƒ
For notations representing multiple numbers, refer to ''Dialing
Plan Notation'' on page 168. However, the asterisk (*) wildcard
cannot be used to depict any source host prefix.
ƒ
If the P-asserted-ID header is present in the incoming INVITE
message, then the parameter 'Source Host Prefix' is compared to
the P-Asserted-ID URI hostname and not to the From header.
Represents a called telephone number prefix.
The prefix can be 1 to 49 digits long.
Note: For notations representing multiple numbers, refer to ''Dialing
Plan Notation'' on page 168.
Source Phone Prefix
[PstnPrefix_SourcePrefix]
Represents a calling telephone number prefix.
The prefix can be 1 to 49 digits long.
Note: For notations representing multiple numbers, refer to ''Dialing
Plan Notation'' on page 168.
Source IP Address
[PstnPrefix_SourceAddress]
The source IP address of an IP-to-Tel call (obtained from the
Contact header in the INVITE message) that can be used for routing
decisions.
Notes:
Version 5.6
ƒ
You can configure from where the source IP address is obtained,
using the parameter SourceIPAddressInput (refer to ''Routing
General Parameters'' on page 171).
ƒ
The source IP address can include the "x" wildcard to represent
single digits. For example: 10.8.8.xx represents all the addresses
between 10.8.8.10 to 10.8.8.99.
ƒ
The source IP address can include the asterisk (*) wildcard to
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Parameter
Description
represent any number between 0 and 255. For example, 10.8.8.*
represents all addresses between 10.8.8.0 and 10.8.8.255.
Trunk Group ID
[PstnPrefix_TrunkGroupId]
The Trunk Group to which incoming SIP calls are assigned that
match all or any combination (including only a single parameter) of
the parameters described above.
Profile ID
[PstnPrefix_ProfileId]
The IP Profile (configured in ''IP Profile Settings'' on page 193) that
is assigned to the routing rule.
Source IP Group ID
[PstnPrefix_SrcIPGroupID]
The source IP Group (1-9) associated with the incoming IP-to-Tel
call. This is the IP Group from where the INVITE message
originated. This IP Group can later be used as the 'Serving IP
Group' in the Account table (refer to ''Configuring the Account Table''
on page 204) for obtaining authentication user name/password for
this call.
3.4.7.4.5 Inbound IP Routing Table
The 'Inbound IP Routing Table' page allows you to identify received calls as inbound IP-toIP calls and assign them to an IP Group (defined in ''Configuring the IP Groups'' on page
201), termed the Source IP Group. This table identifies these IP calls based on any
combination of the following criteria rules:
„
Destination and source host prefixes
„
Destination and source telephone number prefixes
„
Source IP address
Assigning these IP calls to Trunk Group ID '-1' identifies them as inbound IP-to-IP calls.
These calls, now pertaining to an IP Group, can later be routed to an outbound destination
IP Group (refer to ''Outbound IP Routing Table'' on page 178).
Note: The 'Inbound IP Routing Table' page appears only if the parameter
EnableSBC is set to 1 (i.e., enabled) in ''SBC Configuration'' on page 163. If
this parameter is not enabled (default), the 'IP to Trunk Group Routing Table'
page appears instead (refer to ''IP to Trunk Group Routing Table'' on page
181 for a description of this page).
¾ To configure Inbound Routing, take these 5 steps:
1.
Open the 'Inbound IP Routing Table' page (Configuration tab > Protocol
Configuration menu > Routing Tables submenu > IP to Trunk Group Routing page
item).
Figure 3-74: Inbound IP Routing Table
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2.
From the 'Routing Index' drop-down list, select the range of entries that you want to
add.
3.
Configure the table according to the table below.
4.
Click the Submit button to save your changes.
5.
To save the changes so they are available after a power fail, refer to ''Saving
Configuration'' on page 230.
Table 3-47: Inbound IP Routing Table Description
Parameter
IP to Tel Routing Mode
[RouteModeIP2Tel]
Description
Determines whether to route the IP calls before or after manipulation
of the destination number (configured in ''Configuring the Number
Manipulation Tables'' on page 164).
ƒ
[0] Route calls before manipulation = IP-to-IP calls are routed
before the number manipulation rules are applied (default).
ƒ
[1] Route calls after manipulation = IP-to-IP outbound calls are
routed after the number manipulation rules are applied.
Dest. Host Prefix
[PstnPrefix_DestHostPrefix_
The Request URI host name prefix of the incoming SIP INVITE
message. If this routing rule is not required, leave the field empty.
The asterisk (*) symbol can be used to depict any destination host
prefix.
Source Host Prefix
[PstnPrefix_SrcHostPrefix]
The From header URI host name prefix of the incoming INVITE
message. If this routing rule is not required, leave the field empty.
The asterisk (*) symbol can be used to depict any source host
prefix.
Dest. Phone Prefix
[PstnPrefix_DestPrefix]
The called telephone number prefix.
The prefix can be 1 to 49 digits long.
Note: For notations representing multiple numbers, refer to ''Dialing
Plan Notation'' on page 168.
Source Phone Prefix
[PstnPrefix_SourcePrefix]
The calling telephone number prefix.
The prefix can be 1 to 49 digits long.
Note: For notations representing multiple numbers, refer to ''Dialing
Plan Notation'' on page 168.
Source IP Address
[PstnPrefix_SourceAddress]
The source IP address of an IP-to-IP call (obtained from the Contact
header in the SIP INVITE message).
Notes:
ƒ
The source IP address can include the letter "x" wildcard to
represent single digits. For example: 10.8.8.xx represents all the
addresses between 10.8.8.10 to 10.8.8.99.
ƒ
The source IP address can include the asterisk (*) wildcard to
represent any number between 0 and 255. For example, 10.8.8.*
represents all addresses between 10.8.8.0 and 10.8.8.255.
Inbound SIP IP calls matching all or any combination of the above routing rules are subsequently
assigned to the IP Group selected below.
Trunk Group ID
[PstnPrefix_TrunkGroupId]
Identifies these calls as IP-to-IP calls when set to -1.
IP Profile ID
[PstnPrefix_ProfileId]
IP profile (configured in ''IP Profile Settings'' on page 193) assigned
to the inbound IP-to-IP call.
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Parameter
Description
Source IP Group ID
[PstnPrefix_SrcIPGroupID]
The IP Group (1-9) to which you want to assign this inbound IP-to-IP
call. This defines the IP Group (configured in the ''Configuring the IP
Groups'' on page 201) from where the SIP INVITE message is
received. This IP Group can later be used in the 'Outbound IP
Routing' table, and as the Serving IP Group in the ‘Account’ table for
obtaining authentication user name/password for this call.
3.4.7.4.6 Internal DNS Table
The 'Internal DNS Table' page, similar to a DNS resolution is used to translate up to 20 host
(domain) names into IP addresses (e.g., when using the 'Tel to IP Routing' table or
'Outbound IP Routing' table if EnableSBC is enabled). Up to four different IP addresses can
be assigned to the same host name, typically used for alternative routing (for Tel-to-IP call
routing).
Notes:
•
The device initially attempts to resolve a domain name using the Internal
DNS table. If the domain name isn't listed in the table, the device
performs a DNS resolution using an external DNS server.
•
You can also configure the DNS table using the ini file table parameter
DNS2IP (refer to ''Networking Parameters'' on page 260).
¾ To configure the internal DNS table, take these 6 steps:
1.
Open the 'Internal DNS Table' page (Configuration tab > Protocol Configuration
menu > Routing Tables submenu > Internal DNS Table page item).
Figure 3-75: Internal DNS Table Page
2.
In the 'Domain Name' field, enter the host name to be translated. You can enter a
string of up to 31 characters long.
3.
In the 'First IP Address' field, enter the first IP address (in dotted-decimal format
notation) to which the host name is translated.
4.
Optionally, in the 'Second IP Address', 'Third IP Address', and 'Second IP Address'
fields, enter the next IP addresses to which the host name is translated.
5.
Click the Submit button to save your changes.
6.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
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3.4.7.4.7 Internal SRV Table
The 'Internal SRV Table' page provides a table for resolving host names to DNS ARecords. Three different A-Records can be assigned to each host name. Each A-Record
contains the host name, priority, weight, and port.
Notes:
•
If the Internal SRV table is configured, the device initially attempts to
resolve a domain name using this table. If the domain name isn't found,
the device performs an Service Record (SRV) resolution using an
external DNS server.
•
You can also configure the Internal SRV table using the ini file table
parameter SRV2IP (refer to ''Networking Parameters'' on page 260).
¾ To configure the Internal SRV table, take these 9 steps:
1.
Open the 'Internal SRV Table' page (Configuration tab > Protocol Configuration
menu > Routing Tables submenu > Internal SRV Table page item).
Figure 3-76: Internal SRV Table Screen
2.
In the 'Domain Name' field, enter the host name to be translated. You can enter a
string of up to 31 characters long.
3.
From the 'Transport Type' drop-down list, select a transport type.
4.
In the 'DNS Name 1' field, enter the first DNS A-Record to which the host name is
translated.
5.
In the 'Priority', 'Weight' and 'Port' fields, enter the relevant values
6.
Repeat steps 4 through 5, for the second and third DNS names, if required.
7.
Repeat steps 2 through 6, for each entry.
8.
Click the Submit button to save your changes.
9.
To save the changes so they are available after a hardware reset or power fail, refer to
''Saving Configuration'' on page 230.
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3.4.7.4.8 Reasons for Alternative Routing
The 'Reasons for Alternative Routing' page includes two groups - IP to Tel Reasons and Tel
to IP Reasons. Each group allows you to define up to four different release reasons. If a call
is released as a result of one of these reasons, the device tries to find an alternative route
for that call. The release reason for IP-to-Tel calls is provided in Q.931 notation. The
release reason for Tel-to-IP calls is provided in SIP 4xx, 5xx, and 6xx response codes. For
Tel-to-IP calls, an alternative IP address is provided; for IP-to-Tel calls an alternative Trunk
Group is provided. Refer to ''Tel to IP Routing Table'' on page 175 for information on
defining an alternative IP address; refer to ''IP to Trunk Group Routing'' on page 181 for
information on defining an alternative Trunk Group.
You can use the 'Reasons for Alternative Routing' page for the following example
scenarios:
„
Tel-to-IP calls: when there is no response to an INVITE message (after INVITE
retransmissions), the device issues an internal 408 'No Response' implicit release
reason.
„
IP-to-Tel calls: when the destination is busy and release reason #17 is issued or for
other call releases that issue the default release reason (#3). Refer to
DefaultReleaseCause in ''Advanced Parameters'' on page 151.
Notes:
•
The reasons for alternative routing for Tel-to-IP calls only apply when a
Proxy isn't used.
•
For Tel-to-IP calls, the device sends the call to an alternative route only
after the call has failed and the device has subsequently attempted twice
to establish the call unsuccessfully.
•
You can also configure alternative routing using the ini file table
parameters AltRouteCauseTel2IP and AltRouteCauseIP2Tel (refer to
''Number Manipulation and Routing Parameters'' on page 313).
¾ To configure the reasons for alternative routing, take these 5
steps:
1.
Open the 'Reasons for Alternative Routing' page (Configuration tab > Protocol
Configuration menu > Routing Tables submenu > Reasons for Alternative
Routing page item).
Figure 3-77: Reasons for Alternative Routing Page
2.
In the 'IP to Tel Reasons' group, select up to four different call failure reasons that
invoke an alternative IP-to-Tel routing.
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3.
In the 'Tel to IP Reasons' group, select up to four different call failure reasons that
invoke an alternative Tel-to-IP routing.
4.
Click the Submit button to save your changes.
5.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
3.4.7.4.9 Release Cause Mapping
The 'Release Cause Mapping' page consists of two groups that allow the device to map up
to 12 different SIP Responses to Q.850 Release Causes and vice versa, thereby overriding
the hard-coded mapping mechanism (described in ''Release Reason Mapping'' on page
394).
Note: You can also configure SIP Responses-Q.850 Release Causes mapping
using the ini file table parameters CauseMapISDN2SIP and
CauseMapSIP2ISDN (refer to ''ISDN and CAS Interworking-Related
Parameters'' on page 307).
¾ To configure Release Cause Mapping, take these 5 steps:
1.
Open the 'Release Cause Mapping' page (Configuration tab > Protocol
Configuration menu > Routing Tables submenu > Release Cause Mapping page
item).
Figure 3-78: Release Cause Mapping Page
2.
In the 'Release Cause Mapping from ISDN to SIP' group, map (up to 12) different
Q.850 Release Causes to SIP Responses.
3.
In the 'Release Cause Mapping from SIP to ISDN' group, map (up to 12) different SIP
Responses to Q.850 Release Causes.
4.
Click the Submit button to save your changes.
5.
To save the changes so they are available after a power fail, refer to ''Saving
Configuration'' on page 230.
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3.4.7.5
Configuring the Profile Definitions
The Profile Definitions submenu includes the following page items:
„
Coder Group Settings (refer to ''Coder Group Settings'' on page 190)
„
Tel Profile Settings (refer to ''Tel Profile Settings'' on page 192)
„
IP Profile Settings (refer to ''IP Profile Settings'' on page 193)
Implementing the device's Profile features, provides the device with high-level adaptation
when connected to a variety of equipment (at both Tel and IP sides) and protocols, each of
which requires different system behavior.
You can assign different Profiles (behavior) per call, using the call routing tables:
„
'Tel to IP Routing' page (refer to ''Tel to IP Routing Table'' on page 175) or 'Outbound
IP Routing Table' page if EnableSBC is set to 1 (refer to “Outbound IP Routing Table”
on page 178)
„
'IP to Trunk Group Routing' page (refer to ''IP to Trunk Group Routing'' on page 181) or
'Inbound IP Routing Table' page (refer to “Inbound IP Routing Table” on page 184) if
EnableSBC is set to 1,
In addition, you can associate different Profiles per the device's channels.
Each Profile contains a set of parameters such as coders, T.38 Relay, Voice and DTMF
Gain, Silence Suppression, Echo Canceler, RTP DiffServ, Current Disconnect and more.
The Profiles feature allows you to customize these parameters or turn them on or off, per
source or destination routing and/or per the device's trunks (channels). For example,
specific E1/T1 spans can be assigned a Profile that always uses G.711.
Each call can be associated with one or two Profiles - Tel Profile and/or IP Profile. If both IP
and Tel profiles apply to the same call, the coders and other common parameters of the
preferred Profile (determined by the Preference option) are applied to that call. If the
Preference of the Tel and IP Profiles is identical, the Tel Profile parameters take
precedence.
Notes:
•
The default values of the parameters in the 'Tel Profile Settings' and 'IP
Profile Settings' pages are identical to their default values in their
respective primary configuration page.
•
If you modify a parameter in its primary configuration page (orini file) that
also appears in the profile pages, the parameter's new value is
automatically updated in the profile pages. However, once you modify
any parameter in the profile pages, modifications to parameters in the
primary configuration pages (orini file) no longer impact that profile
pages.
3.4.7.5.1 Coder Group Settings
The 'Coder Group Settings' page provides a table for defining up to four different coder
groups. These coder groups are used in the 'Tel Profile Settings' and 'IP Profile Settings'
pages to assign different coders to Profiles.
For each coder group you can define up to five coders, where the first coder (and its
attributes) in the table takes precedence over the second coder, and so on. The first coder
is the highest priority coder and is used by the device whenever possible. If the far end
device cannot use the coder assigned as the first coder, the device attempts to use the next
coder and so on. For a list of coders supported by the device, refer to ''Coders'' on page
144.
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Notes:
•
Each coder type can appear only once per Coder Group.
•
The device always uses the packetization time requested by the remote
side for sending RTP packets. If not specified, the packetization time
(ptime) is assigned the default value.
•
Only the packetization time of the first coder in the defined coder list is
declared in INVITE / 200 OK SDP, even if multiple coders are defined.
•
For G.729, you can also select silence suppression without adaptations.
•
If silence suppression is enabled for G.729, the device includes the string
'annexb=no' in the SDP of the relevant SIP messages. If silence
suppression is set to 'Enable w/o Adaptations', 'annexb=yes' is included.
An exception is when the remote device is a Cisco gateway
(IsCiscoSCEMode).
•
You can also configure the coder groups using the ini file table parameter
CoderName (refer to ''SIP Configuration Parameters'' on page 284).
¾ To configure coder groups, take these 11 steps:
1.
Open the 'Coder Group Settings' page (Configuration tab > Protocol Configuration
menu > Profile Definitions submenu > Coder Group Settings page item).
Figure 3-79: Coder Group Settings Page
2.
From the 'Coder Group ID' drop-down list, select a coder group ID.
3.
From the 'Coder Name' drop-down list, select the first coder for the coder group.
4.
From the 'Packetization Time' drop-down list, select the packetization time (in msec)
for the coder. The packetization time determines how many coder payloads are
combined into a single RTP packet.
5.
From the 'Rate' drop-down list, select the bit rate (in kbps) for the coder you selected.
6.
In the 'Payload Type' field, if the payload type for the coder you selected is dynamic,
enter a value from 0 to 120 (payload types of 'well-known' coders cannot be modified).
The payload type identifies the format of the RTP payload.
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7.
From the 'Silence Suppression' drop-down list, enable or disable the silence
suppression option for the coder you selected.
8.
Repeat steps 3 through 7 for the second to fifth coders (optional).
9.
Repeat steps 2 through 8 for the second to fourth coder groups (optional).
10. Click the Submit button to save your changes.
11. To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
3.4.7.5.2 Tel Profile Settings
The 'Tel Profile Settings' page allows you to define up to nine different Tel Profiles. You can
then assign these Tel Profiles to the device's channels (in the 'Trunk Group Table' page),
thereby applying different behaviors to different channels.
Note: You can also configure Tel Profiles using the ini file table parameter TelProfile
(refer to ''SIP Configuration Parameters'' on page 284).
¾ To configure Tel Profiles, take these 9 steps:
1.
Open the 'Tel Profile Settings' page (Configuration tab > Protocol Configuration
menu > Profile Definitions submenu > Tel Profile Settings page item).
Figure 3-80: Tel Profile Settings Page
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2.
From the 'Profile ID' drop-down list, select the Tel Profile identification number you
want to configure.
3.
In the 'Profile Name' field, enter an arbitrary name that enables you to easily identify
the Tel Profile.
4.
From the 'Profile Preference' drop-down list, select the priority of the Tel Profile, where
'1' is the lowest priority and '20' is the highest. If both IP and Tel profiles apply to the
same call, the coders and other common parameters (noted by an asterisk in the
description of the parameter TelProfile) of the preferred Profile are applied to that call.
If the Preference of the Tel and IP Profiles is identical, the Tel Profile parameters are
applied.
Note: If the coder lists of both IP and Tel Profiles apply to the same call, only the
coders common to both are used. The order of the coders is determined by the
preference.
5.
Configure the Profile's parameters according to your requirements. For detailed
information on each parameter, refer to its description on the page in which it is
configured as an individual parameter.
6.
From the 'Coder Group' drop-down list, select the Coder Group (refer to ''Coder Group
Settings'' on page 190) or the device's default coder (refer to ''Coders'' on page 144) to
which you want to assign the Profile.
7.
Repeat steps 2 through 6 to configure additional Tel Profiles (optional).
8.
Click the Submit button to save your changes.
9.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
3.4.7.5.3 IP Profile Settings
The 'IP Profile Settings' page allows you to define up to nine different IP Profiles. You can
then assign these IP Profiles to routing rules in the 'Tel to IP Routing' page (refer to ''Tel to
IP Routing Table'' on page 175) or 'Outbound IP Routing Table' if EnableSBC is set to 1
(refer to “Outbound IP Routing Table” on page 178) and 'IP to Trunk Group Routing' page
(refer to ''IP to Trunk Group Routing'' on page 181) or 'Inbound IP Routing Table' if
EnableSBC is set to 1 (refer to “Inbound IP Routing Table” on page 184). IP Profiles can
also be used when working with a Proxy server (set AlwaysUseRouteTable to 1).
Note: You can also configure the IP Profiles using the ini file table parameter
IPProfile (refer to ''SIP Configuration Parameters'' on page 284).
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¾ To configure the IP Profile settings, take these 9 steps:
1.
Open the 'IP Profile Settings' page (Configuration tab > Protocol Configuration
menu > Profile Definitions submenu > IP Profile Settings page item).
Figure 3-81: IP Profile Settings Page
2.
From the 'Profile ID' drop-down list, select an identification number for the IP Profile.
3.
In the 'Profile Name' field, enter an arbitrary name that allows you to easily identify the
IP Profile.
4.
From the 'Profile Preference' drop-down list, select the priority of the IP Profile, where
'1' is the lowest priority and '20' is the highest. If both IP and Tel profiles apply to the
same call, the coders and other common parameters (noted by an asterisk in the
description of the parameter IPProfile) of the preferred Profile are applied to that call. If
the Preference of the Tel and IP Profiles is identical, the Tel Profile parameters are
applied.
Note: If the coder lists of both IP and Tel Profiles apply to the same call, only the
coders common to both are used. The order of the coders is determined by the
preference.
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5.
Configure the IP Profile's parameters according to your requirements. For detailed
information on each parameter, refer to the description on the page in which it is
configured as an individual parameter. Parameters that are unique to IP Profile are
described in the table below.
6.
From the 'Coder Group' drop-down list, select the coder group you want to assign to
the Profile. You can select the device's default coders (refer to ''Coders'' on page 144)
or one of the coder groups you defined in the 'Coder Group Settings' page (refer to
''Coder Group Settings'' on page 190).
7.
Repeat steps 2 through 6 for the next IP Profiles (optional).
8.
Click the Submit button to save your changes.
9.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-48: Description of Parameter Unique to IP Profile
Parameter
Number of Calls Limit
3.4.7.6
Description
Maximum number of concurrent calls. If the profile is set to some limit, the
device maintains the number of concurrent calls (incoming and outgoing)
pertaining to the specific profile. A limit value of '-1' indicates that there is
no limitation on calls for that specific profile (default). A limit value of '0'
indicates that all calls are rejected. When the number of concurrent calls is
equal to the limit, the device rejects any new incoming and outgoing calls
belonging to that profile.
Configuring the Trunk and IP Groups
The Trunk/IP Group menu allows you to configure groups of channels. This submenu
includes the following page items:
„
Trunk Group (refer to “Configuring the Trunk Group Table” on page 195)
„
Trunk Group Settings (refer to ''Configuring the Trunk Group Settings'' on page 197)
„
IP Group Table (refer to ''Configuring the IP Groups'' on page 201)
„
Account Table (refer to ''Configuring the Account Table'' on page 204)
3.4.7.6.1 Configuring the Trunk Group Table
The 'Trunk Group Table' page provides you with a table for enabling device channels, by
assigning them telephone numbers, Trunk Groups, and Profiles. Trunk Groups are used for
routing IP-to-Tel calls with common rules. Channels that are not defined are disabled. You
can add up to 24entries in this table.
Note: You can also configure the Trunk Groups using the ini file table parameter
TrunkGroup_x to (refer to ''Number Manipulation and Routing Parameters'' on
page 313).
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¾ To configure the Trunk Group table, take these 4 steps:
1.
Open the 'Trunk Group Table' page (Configuration tab > Protocol Configuration
menu > Trunk/IP Group submenu > Trunk Group page item).
Figure 3-82: Trunk Group Table Page
2.
Configure the Trunk Group according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to the flash memory, refer to ''Saving Configuration'' on page
230.
Table 3-49: Trunk Group Table Description
Parameter
Description
From Trunk
[TrunkGroup_FirstTrunkId]
Starting physical Trunk number. The number of listed Trunks
depends on the device's hardware configuration.
To Trunk
[TrunkGroup_LastTrunkId]
Ending physical Trunk number. The number of listed Trunks
depends on the device's hardware configuration.
Channels
[TrunkGroup_FirstBChannel],
[TrunkGroup_LastBChannel]
The device's Trunk B-channels. To enable the channels, enter
the channel numbers. You can enter a range of channels by
using the format [n-m], where n represents the lower channel
number and m the higher channel number, e.g., [1-24]
specifies channels 1 through 24.
Notes:
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The number of defined channels must not exceed the
number of the Trunk’s B-channels.
ƒ
To represent all channels, enter a single asterisk (*).
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Parameter
Description
Phone Number
[TrunkGroup_FirstPhoneNumber]
Enter the first telephone number that you want to assign to the
first channel defined in the 'Channels' field. Subsequent
channels are assigned the next consecutive phone number.
Notes:
Trunk Group ID
[TrunkGroup_TrunkGroupNum]
ƒ
If the 'Phone Number' field includes alphabetical characters
and the phone number is defined for a range of channels
(e.g., 1-4), then the phone number must end with a number
(e.g., 'user1').
ƒ
This field is optional. The logical numbers defined in this
field are used when an incoming PSTN / PBX call doesn't
contain the calling number or called number (the latter
being determined by the parameter
ReplaceEmptyDstWithPortNumber); these numbers are
used to replace them. These logical numbers are also used
for channel allocation for IP-to-Tel calls if the Trunk Group's
'Channel Select Mode' is set to 'By Dest Phone Number'.
The Trunk Group ID (1-99) assigned to the corresponding
channels. The same Trunk Group ID can be used for more
than one group of channels. Trunk Group ID is used to define
a group of common channel behavior that are used for routing
IP-to-Tel calls. If an IP-to-Tel call is assigned to a Trunk
Group, the call is routed to the channel or channels that
correspond to the Trunk Group ID.
You can configure the 'Trunk Group Settings' table (refer to
''Configuring the Trunk Group Settings'' on page 197) to
determine the method in which new calls are assigned to
channels within the Trunk Groups.
Note: You must configure the 'IP to Trunk Group Routing
Table' page (refer to ''IP to Trunk Group Routing'' on page 181)
to assign incoming IP calls to the appropriate Trunk Group. If
you do not configure the 'IP to Trunk Group Routing Table',
calls do not complete.
Profile ID
[TrunkGroup_ProfileId]
The Tel profile ID (refer to ''Tel Profile Settings'' on page 192)
assigned to the channels defined in the 'Channels' field.
3.4.7.6.2 Configuring the Trunk Group Settings
The 'Trunk Group Settings' page is mainly used to select the method for which IP-to-Tel
calls are assigned to channels within each Trunk Group. If no method is selected (for a
specific Trunk Group), the setting of the global parameter ChannelSelectMode in the 'SIP
General Parameters' page (refer to ''SIP General Parameters'' on page 121) applies. In
addition, this page also defines the method for registering Trunk Groups to selected Serving
IP Group IDs (if defined). You can add up to 24 entries in this table.
Note: You can also configure the Trunk Group Settings table using the ini file table
parameter TrunkGroupSettings (refer to ''Number Manipulation and Routing
Parameters'' on page 313).
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¾ To configure the Trunk Group Settings table, take these 5 steps:
1.
Open the 'Trunk Group Settings' page (Configuration tab > Protocol Configuration
menu > Trunk/IP Group submenu > Trunk Group Settings page item).
Figure 3-83: Trunk Group Settings Page
2.
From the 'Routing Index' drop-down list, select the range of entries that you want to
edit (up to 24 entries can be configured).
3.
Configure the Trunk Group according to the table below.
4.
Click the Submit button to save your changes.
5.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-50: Trunk Group Settings Parameters Description
Parameter
Description
Trunk Group ID
[TrunkGroupSettings_Trun
kGroupId]
The Trunk Group ID that you want to configure.
The valid range is 1-99.
Trunks are assigned to Trunk Groups in the 'Trunk Group Table'
page (refer to “Configuring the Trunk Group Table” on page 195).
Channel Select Mode
[TrunkGroupSettings_Chan
nelSelectMode]
The method in which IP-to-Tel calls are assigned to channels
pertaining to a Trunk Group:
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[0] By Dest Phone Number = Selects the device's channel
according to the called number defined in the 'Trunk Group Table'
(refer to “Configuring the Trunk Group Table” on page 195).
ƒ
[1] Cyclic Ascending (default) = Selects the next available
channel in an ascending cyclic order. The next highest channel
number in the Trunk Group is always selected. When the highest
channel number in the Trunk Group is reached, the lowest
channel number in the Trunk Group is selected, and then it starts
ascending again.
ƒ
[2] Ascending = Selects the lowest available channel. The lowest
channel number in the Trunk Group is always first selected, and if
that channel is unavailable, the next highest channel is selected.
ƒ
[3] Cyclic Descending = Selects the next available channel in
descending cyclic order. The next lowest channel number in the
Trunk Group is always first selected. When the lowest channel
number in the Trunk Group is reached, it selects the highest
channel number in the Trunk Group and then start descending
again.
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Parameter
Registration Mode
[TrunkGroupSettings_Regis
trationMode]
Description
ƒ
[4] Descending = Selects the highest available channel. The
highest channel number in the Trunk Group is always first
selected, and if that channel is unavailable, the next lowest
channel is selected.
ƒ
[5] Dest Number + Cyclic Ascending = The channel is first
selected according to the called number. If the called number isn't
found, the next available channel in ascending cyclic order is
selected. Note that if the called number is found, but the channel
associated with the number is busy, the call is released.
ƒ
[6] By Source Phone Number = Selects the channel according to
the calling number.
ƒ
[7] Trunk Cyclic Ascending = The first channel of the next Trunk
(i.e., next to the Trunk from which the previous channel was
allocated) is selected.
Registration mode per Trunk Group:
ƒ
[1] Per Gateway = Single registration for the entire device
(default). This mode is applicable only if a default Proxy or
Registrar IP are configured, and Registration is enabled (i.e.,
parameter IsRegisterUsed is set to 1). In this mode, the URI
userpart in the From, To, and Contact headers is set to the value
of the global registration parameter GWRegistrationName (refer
to ''Proxy & Registration Parameters'' on page 132) or username
if GWRegistrationName is not configured.
ƒ
[0] Per Endpoint = Each channel in the Trunk Group registers
individually. The registrations are sent to the ServingIPGroupID if
defined in the table, otherwise to the default Proxy, and if no
default Proxy, then to the Registrar IP.
ƒ
[4] Don't Register = No registrations are sent by endpoints
pertaining to the Trunk Group. For example, if the device is
configured globally to register all its endpoints (using the
parameter ChannelSelectMode), you can exclude some
endpoints from being registered by assigning them to a Trunk
Group and configuring the Trunk Group registration mode to
'Don't Register'.
ƒ
[5] Per Account = Registrations are sent (or not) to an IP Group,
according to the settings in the Account table (refer to
''Configuring the Account Table'' on page 204).
Notes:
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ƒ
To enable Trunk Group registrations, configure the global
parameter IsRegisterNeeded to 1. This is unnecessary for 'Per
Account' registration mode.
ƒ
If no mode is selected, the registration is performed according to
the global registration parameter ChannelSelectMode (refer to
''Proxy & Registration Parameters'' on page 132).
ƒ
If the device is configured globally (ChannelSelectMode) to
register Per Endpoint, and a Trunk Group comprising four
channels is configured to register Per Gateway, the device
registers all channels except the first four channels. The Trunk
Group of these four channels sends a single registration request.
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Parameter
Description
Serving IP Group ID
[TrunkGroupSettings_Servi
ngIPGroup]
The Serving IP Group ID to where INVITE messages initiated by this
Trunk Group's endpoints are sent. The actual destination to where
these INVITE messages are sent is to the Proxy Set ID (refer to
''Proxy Sets Table'' on page 141) associated with this Serving IP
Group. The Request URI hostname in the INVITE and REGISTER
messages (except for 'Per Account' registration modes) is set to the
value of the field 'SIP Group Name' defined in the 'IP Group' table
(refer to ''Configuring the IP Groups'' on page 201).
If no Serving IP Group ID is selected, the INVITE messages are sent
to the default Proxy or according to the 'Tel to IP Routing Table'
(refer to ''Tel to IP Routing Table'' on page 175) or 'Outbound IP
Routing Table' if EnableSBC is set to 1 (refer to “Outbound IP
Routing Table” on page 178).
Note: If the parameter PreferRouteTable is set to 1 (refer to ''Proxy
& Registration Parameters'' on page 132), the routing rules in the
'Tel to IP Routing Table' (or 'Outbound IP Routing Table') prevail
over the selected Serving IP Group ID.
Gateway Name
[TrunkGroupSettings_Gate
wayName]
The host name used in the From header in INVITE messages, and
as a host name in From/To headers in REGISTER requests. If not
configured, the global parameter SIPGatewayName is used instead.
Contact User
[TrunkGroupSettings_Cont
actUser]
This is used as the user part in the Contact URI in INVITE
messages, and as a user part in From, To, and Contact headers in
REGISTER requests. This is applicable only if the field 'Registration
Mode' is set to 'Per Account', and the Registration through the
Account table is successful.
Notes:
ƒ
If registration fails, then the userpart in the INVITE Contact
header contains the source party number.
ƒ
The 'ContactUser' parameter in the 'Account Table' page
overrides this parameter.
An example is shown below of a REGISTER message for registering endpoint "101" using registration
Per Endpoint mode. The "SipGroupName" in the request URI is taken from the IP Group table.
REGISTER sip:SipGroupName SIP/2.0
Via: SIP/2.0/UDP 10.33.37.78;branch=z9hG4bKac862428454
From: <sip:101@GatewayName>;tag=1c862422082
To: <sip:101@GatewayName>
Call-ID: [email protected]
CSeq: 3 REGISTER
Contact: <sip:[email protected]>;expires=3600
Expires: 3600
User-Agent: Audiocodes-Sip-Gateway/v.5.40A.008.002
Content-Length: 0
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3.4.7.6.3 Configuring the IP Groups
The 'IP Group Table' page allows you to create up to nine logical IP entities (IP Groups)
that are later used in the call routing tables. The IP Groups are typically implemented in Telto-IP call routing. The IP Group can be used as a destination entity in the 'Tel to IP Routing'
table (or 'Outbound IP Routing Table'), and Serving IP Group ID in the 'Trunk Group
Settings' (refer to ''Configuring the Trunk Group Settings'' on page 197) and 'Account' (refer
to ''Configuring the Account Table'' on page 204) tables. These call routing tables are used
for identifying the IP Group from where the INVITE is sent for obtaining a digest
user/password from the 'Account' table if there is a need to authenticate subsequent SIP
requests in the call. The IP Group can also be implemented in IP-to-Tel call routing (or
inbound IP routing) as a source IP Group.
The IP Groups are assigned various entities such as a Proxy Set ID, which represents an
IP address (created in ''Proxy Sets Table'' on page 141). You can also assign the IP Group
with a host name and other parameters that reflect parameters sent in SIP Request
From\To.
Notes:
•
By default, if you disable the use of a proxy (i.e., IsProxyUsed is set to 0),
then only one IP Group is defined (and working with multiple IP Groups is
not valid).
•
You can also configure the IP Groups table using the ini file table
parameter IPGroup (refer to ''SIP Configuration Parameters'' on page
284).
¾ To configure IP Groups, take these 4 steps:
1.
Open the 'IP Group Table' page (Configuration tab > Protocol Configuration menu
> Trunk/IP Group submenu > IP Group Table page item).
Figure 3-84: IP Group Table Page
2.
Configure the IP group parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
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Table 3-51: IP Group Parameters Description
Parameter
Type
Description
The IP Group can be defined as one of the following types:
ƒ
SERVER = used when the destination address (configured by the
Proxy Set) of the IP Group (e.g., ITSP, Proxy, IP-PBX, or Application
server) is known.
ƒ
USER = represents a group of users (such as IP phones and
softphones) where their location is dynamically obtained by the device
when REGISTER requests and responses traverse (or are terminated)
by the device. These users are considered remote (far-end) users.
Typically, this IP Group is configured with a Serving IP Group that
represents an IP-PBX, Application or Proxy server that serves this
USER-type IP Group. Each SIP request sent by a user of this IP
Group is proxied to the Serving IP Group. For registrations, the device
updates its internal database with the AOR and contacts of the users.
Digest authentication using SIP 401/407 responses (if needed) is
performed by the Serving IP Group. The device forwards these
responses directly to the SIP users.
To route a call to a registered user, a rule must be configured in the
‘Outbound IP Routing’ table (refer to “Outbound IP Routing Table” on
page 178). The device searches the dynamic database (by using the
request URI) for an entry that matches a registered AOR or Contact.
Once an entry is found, the IP destination is obtained from this entry,
and a SIP request is sent to the destination.
The device also supports NAT traversal for the SIP clients that are behind
NAT. In this case, the device must be defined with a global IP address.
Note: This field is available only if EnableSBC is set to 1 (refer to “SBC
Configuration” on page 163).
Description
Brief string description of the IP Group.
The value range is a string of up to 29 characters. The default is an empty
field.
Proxy Set ID
Selects the Proxy Set ID (defined in ''Proxy Sets Table'' on page 141) to
associate with the IP Group. All INVITE messages configured to be 'sent'
to the specific IP Group are in fact sent to the IP address associated with
this Proxy Set.
The range is 0-5, where 0 is the default Proxy Set.
Note: The Proxy Set is only defined for SERVER type IP Groups.
SIP Group Name
The request URI host name used in INVITE and REGISTER messages
that are sent to this IP Group, or the host name in the From header of
INVITE messages received from this IP Group. If not specified, the value
of the global parameter ProxyName (refer to ''Proxy & Registration
Parameters'' on page 132) is used instead.
The value range is a string of up to 49 characters. The default is an empty
field.
Note: If the IP Group is of type USER, this parameter is used internally as
a hostname in the request URI for TDM-to-IP initiated calls. For example,
if an incoming call from the device's T1 trunk is routed to a USER-type IP
Group, the device first forms the request URI
(destination_number@SIPGroupName), and then it searches the User’s
internal database for a match.
Contact User
SIP User's Manual
Defines the user part for the From, To, and Contact headers of SIP
REGISTER messages, and the user part for the Contact header of
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Parameter
Description
INVITE messages that are received from this IP Group and forwarded by
the device to another IP Group.
Notes:
Serving IP Group ID
Enable Survivability
ƒ
This parameter is applicable only for USER-type IP Groups.
ƒ
This parameter is overridden by the ‘Contact User’ parameter (if
configured) in the ‘Account’ table (refer to ''Configuring the Account
Table'' on page 204).
If configured, INVITE messages initiated from the IP Group are sent to
this Serving IP Group (range 1 to 9). In other words, the INVITEs are sent
to the address defined for the Proxy Set associated with this Serving IP
Group. The Request URI host name in these INVITE messages are set to
the value of the parameter ‘SIP Group Name’ defined for the Serving IP
Group.
Notes:
ƒ
This field is available only if EnableSBC is set to 1 (refer to “SBC
Configuration” on page 163).
ƒ
If the parameter PreferRouteTable is set to 1, the routing rules in the
‘Outbound IP Routing’ table takes precedence over this ‘Serving IP
Group ID’ parameter.
ƒ
If this parameter is not configured, the INVITE messages are sent to
the default Proxy or according to the ‘Outbound IP Routing’ table.
Determines whether Survivability mode is enabled for USER-type IP
Groups.
ƒ
Disable (default).
ƒ
Enable = Survivability mode is enabled. The device records in its local
database the registration messages sent by the clients belonging to
the USER-type IP Group. If communication with the Serving IP Group
(e.g., IP-PBX) fails, the USER-type IP Group enters into Survivability
mode in which the device uses its database for routing calls between
the clients (e.g., IP phones) of the USER-type IP Group. The RTP
packets between the IP phones in Survivability mode always traverse
through the device. In Survivability mode, the device is capable of
receiving new registrations. When the Serving IP Group is available
again, the device returns to normal mode, sending INVITE and
REGISTER messages to the Serving IP Group.
Notes:
Routing Mode
Version 5.6
ƒ
This field is available only if EnableSBC is set to 1 (refer to “SBC
Configuration” on page 163).
ƒ
This parameter is applicable only to USER-type IP Groups.
Defines the routing mode for outgoing SIP INVITE messages.
ƒ
[0] Not Configured = The routing is done according to the selected
Serving IP Group. If no Serving IP Group is selected, the device routes
the call according to the 'Outbound IP Routing' table (refer to
“Outbound IP Routing Table” on page 178).
ƒ
[1] Routing Table = The device routes the call according to the
'Outbound IP Routing' table.
ƒ
[2] Serving IP Group = The device sends the SIP INVITE to the
selected Serving IP Group. If no Serving IP Group is selected, the
default IP Group is used. If the Proxy server(s) associated with the
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Parameter
Description
destination IP Group is not alive, the device uses the 'Outbound IP
Routing' table (if the parameter IsFallbackUsed is set 1, i.e., fallback
enabled - refer to Proxy & Registration Parameters on page 132).
ƒ
[3] Request-URI = The device sends the SIP INVITE to the IP address
according to the received SIP Request-URI host name.
Note: This field is available only if EnableSBC is set to 1 (refer to “SBC
Configuration” on page 163).
SIP Re Routing Mode
Determines the routing mode after a call redirection (i.e., a 3xx SIP
response is received) or transfer (i.e., a SIP REFER request is received).
ƒ
[0] Standard = INVITE messages that are generated as a result of
Transfer or Redirect are sent directly to the URI, according to the
Refer-To header in the REFER message or Contact header in the 3xx
response (default).
ƒ
[1] Proxy = Sends a new INVITE to the Proxy. Note: Applicable only if
a Proxy server is used and the parameter AlwaysSendtoProxy is set to
0.
ƒ
[2] Routing Table = Uses the Routing table to locate the destination
and then sends a new INVITE to this destination.
Notes:
Always Use Route Table
ƒ
When this parameter is set to [1] and the INVITE sent to the Proxy
fails, the device re-routes the call according to the Standard mode [0].
ƒ
When this parameter is set to [2] and the INVITE fails, the device reroutes the call according to the Standard mode [0]. If DNS resolution
fails, the device attempts to route the call to the Proxy. If routing to the
Proxy also fails, the Redirect / Transfer request is rejected.
ƒ
When this parameter is set to [2], the XferPrefix parameter can be
used to define different routing rules for redirected calls.
ƒ
This parameter is disregarded if the parameter AlwaysSendToProxy is
set to 1.
Determines the Request URI host name in outgoing INVITE messages.
ƒ
Disable (default).
ƒ
Enable = The device uses the IP address (or domain name) defined in
the 'Tel to IP Routing' table (''Tel to IP Routing Table'' on page 175) as
the Request URI host name in outgoing INVITE messages, instead of
the value entered in the 'SIP Group Name' field.
3.4.7.6.4 Configuring the Account Table
The 'Account Table' page allows you to define accounts per Trunk Groups (referred to as
Served Trunk Group) or to a Served IP Group for registration and/or digest authentication
(user name and password) to a destination IP address (Serving IP Group). The Account
table can be used, for example, to register to an Internet Telephony Service Provider (ITSP)
on behalf of an IP-PBX to which the device is connected. The registrations are sent to the
Proxy Set ID (refer to ''Proxy Sets Table'' on page 141) associated with these Serving IP
Groups. A Trunk Group can register to more than one Serving IP Group (e.g., ITSP's), by
configuring multiple entries in this Account table with the same Served Trunk Group, but
with different Serving IP Groups, user name/password, Host Name, and Contact User
parameters.
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Note: You can also configure the Account table using the ini file table parameter
Account (refer to ''SIP Configuration Parameters'' on page 284).
¾ To configure Accounts, take these 5 steps:
1.
Open the 'Account Table' page (Configuration tab > Protocol Configuration menu >
Trunk/IP Group submenu > Account Table page item).
Figure 3-85: Account Table Page
2.
To add an Account, in the 'Add' field, enter the desired table row index, and then click
Add. A new row appears.
3.
Configure the Account parameters according to the table below.
4.
Click the Apply button to save your changes.
5.
To save the changes, refer to ''Saving Configuration'' on page 230.
Note: For a description of the Web interface's table command buttons (e.g.,
Duplicate and Delete), refer to ''Working with Tables'' on page 30.
Table 3-52: Account Parameters Description
Parameter
Served Trunk
Group
Description
The Trunk Group ID for which the device performs registration and/or
authentication to a destination IP Group (i.e., Serving IP Group).
For Tel-to-IP calls, the Served Trunk Group is the source Trunk Group from
where the call initiated. For IP-to-Tel calls, the Served Trunk Group is the
'Trunk Group ID' defined in the 'IP to Trunk Group Routing' table (refer to ''IP to
Trunk Group Routing'' on page 181). For defining Trunk Groups, refer to
“Configuring the Trunk Group Table” on page 195 .
Note: For IP-to-IP call routing, this parameter must be set to -1 (i.e., no trunk).
Served IP Group
The Source IP Group (e.g., IP-PBX) for which registration and/or authentication
is performed.
Serving IP Group
The destination IP Group ID (defined in ''Configuring the IP Groups'' on page
201) to where the REGISTER requests (if enabled) are sent or Authentication
is performed. The actual destination to where the REGISTER requests are sent
is the IP address defined for the Proxy Set ID (refer to ''Proxy Sets Table'' on
page 141) associated with this IP Group. This occurs only in the following
conditions:
ƒ
Version 5.6
The parameter 'Registration Mode' is set to 'Per Account' in the 'Trunk
Group Settings' table (refer to ''Configuring the Trunk Group Settings'' on
page 197).
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Parameter
Description
ƒ
The parameter 'Register' in this table is set to 1.
In addition, for a SIP call that is identified by both the Served Trunk Group/
Served IP Group and Serving IP Group, the username and password for digest
authentication defined in this table is used.
For Tel-to-IP calls, the Serving IP Group is the destination IP Group defined in
the 'Trunk Group Settings' table or 'Tel to IP Routing' table (refer to ''Tel to IP
Routing Table'' on page 175). For IP-to-Tel calls, the Serving IP Group is the
'Source IP Group ID' defined in the 'IP to Trunk Group Routing' table (refer to
''IP to Trunk Group Routing'' on page 181).
Note: If no match is found in this table for incoming or outgoing calls, the
username and passwordthe global parameters (UserName and Password)
defined on the 'Proxy & Registration' page (refer to ''Proxy & Registration
Parameters'' on page 132) are used.
Username
Digest MD5 Authentication user name (up to 50 characters).
Password
Digest MD5 Authentication password (up to 50 characters).
HostName
Defines the Address of Record (AOR) host name. It appears in REGISTER
From/To headers as ContactUser@HostName. For successful registrations,
this HostName is also included in the INVITE request's From header URI. If not
configured or if registration fails, the 'SIP Group Name' parameter from the ‘IP
Group’ table is used instead.
This parameter can be up to 49 characters.
Register
Enables registration.
ƒ
No = Don't register
ƒ
Yes = Register
When enabled, the device sends REGISTER requests to the Serving IP Group.
In addition, to activate registration, you also need to set the parameter
'Registration Mode' to 'Per Account' in the 'Trunk Group Settings' table (refer to
''Configuring the Trunk Group Settings'' on page 197) for the specific Trunk
Group. The Host Name (i.e., host name in SIP From/To headers) and Contact
User (user in From/To and Contact headers) are taken from this table upon a
successful registration. See the example below:
REGISTER sip:audiocodes SIP/2.0
Via: SIP/2.0/UDP 10.33.37.78;branch=z9hG4bKac1397582418
From: <sip:ContactUser@HostName>;tag=1c1397576231
To: <sip: ContactUser@HostName >
Call-ID: [email protected]
CSeq: 1 REGISTER
Contact: <sip:[email protected]>;expires=3600
Expires: 3600
User-Agent: Audiocodes-Sip-Gateway/v.5.40A.008.002
Content-Length: 0
Notes:
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ƒ
The Trunk Group account registration is not effected by the parameter
IsRegisterNeeded.
ƒ
You can configure this table so that a specific IP Group can register to
multiple ITSP’s.This is performed by defining several rows in this table
containing the same Served IP Group, but with different Serving IP Groups,
user/password, Host Name and Contact User parameters.
ƒ
If registration to an IP Group(s) fails for all the accounts defined in this table
for a specific Trunk Group, and if this Trunk Group includes all the channels
in the Trunk (refer to “Configuring the Trunk Group Table” on page 195), the
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Parameter
Description
Trunk Group is set to Out-Of-Service if the parameter
OOSOnRegistrationFail is set to 1 (refer to ''Proxy & Registration
Parameters'' on page 132).
Contact User
Defines the AOR user name. It appears in REGISTER From/To headers as
ContactUser@HostName, and in INVITE/200 OK Contact headers as
ContactUser@<device's IP address>. If not configured, the 'Contact User'
parameter from the 'IP Group Table' page is used instead.
Note: If registration fails, then the userpart in the INVITE Contact header
contains the source party number.
3.4.7.7
Configuring the Digital Gateway Parameters
The 'Digital Gateway Parameters' page allows you to configure miscellaneous digital
parameters.
¾ To configure the digital gateway parameters, take these 4 steps:
1.
Open the 'Digital Gateway Parameters' page (Configuration tab > Protocol
Configuration menu > Digital Gateway submenu > Digital Gateway Parameters
page item).
Figure 3-86: Digital Gateway Parameters Page
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2.
Configure the Digital Gateway parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-53: Digital Gateway Parameters Description
Parameter
B-channel Negotiation
[BchannelNegotiation]
Description
Determines the ISDN B-Channel negotiation mode.
ƒ
[0] Preferred.
ƒ
[1] Exclusive (default).
ƒ
[2] Any.
Notes:
ƒ
Applicable only to ISDN protocols.
ƒ
For some ISDN variants, when 'Any' (2) is selected, the
SETUP message does not include the Channel Identification
IE.
ƒ
The 'Any' (2) option is applicable only if the parameter 'ISDN
Termination Side' is set to 'Use side' (refer to ''Configuring the
Trunk Settings'' on page 82).
Swap Redirect and Called
Numbers
[SwapRedirectNumber]
ƒ
[0] No = Don't change numbers (default).
ƒ
[1] Yes = Incoming ISDN call that includes a redirect number
(sometimes referred to as 'original called number') uses the
redirect number instead of the called number.
MFC R2 Category
[R2Category]
Determines the tone for MFC R2 Calling Party Category (CPC).
The parameter provides information on the calling party such as
National or International call, Operator or Subscriber and
Subscriber priority.
The value range is 1 to 15 (defining one of the MFC R2 tones).
The default value is 1.
Disconnect Call on Busy Tone
Detection (CAS)
[DisconnectOnBusyTone]
Determines whether a call is disconnected upon detection of a
busy tone.
ƒ
[0] Disable = Do not disconnect call on detection of busy tone.
ƒ
[1] Enable = Disconnect call on detection of busy tone
(default).
Note: This parameter is applicable only to CAS protocols.
Disconnect Call on Busy Tone
Detection (ISDN)
[ISDNDisconnectOnBusyTone]
Determines whether a call is disconnected upon detection of a
busy tone.
ƒ
[0] = Do not disconnect call upon detection of busy tone.
ƒ
[1] = Disconnect call upon detection of busy tone (default).
Note: This parameter is applicable only to ISDN protocols.
Enable TDM Tunneling
[EnableTDMoverIP]
Enables TDM tunneling.
ƒ
[0] Disable = Disabled (default).
ƒ
[1] Enable = TDM Tunneling is enabled.
When TDM Tunneling is enabled, the originating device
automatically initiates SIP calls from all enabled B-channels
pertaining to E1/T1/J1 spans that are configured with the
'Transparent' protocol. The called number of each call is the
internal phone number of the B-channel from where the call
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Parameter
Description
originates. The 'IP to Trunk Group' routing table is used to define
the destination IP address of the terminating device. The
terminating device automatically answers these calls if its E1/T1
protocol is set to 'Transparent' (ProtocolType = 5).
Send Screening Indicator to IP
[ScreeningInd2IP]
Overrides the calling party's number (CPN) screening indication
in the received ISDN SETUP message for Tel-to-IP calls.
ƒ
[-1] Not Configured = not configured (interworking from ISDN
to IP) or set to 0 for CAS (default).
ƒ
[0] User Provided = CPN set by user, but not screened
(verified).
ƒ
[1] User Passed = CPN set by user, verified and passed.
ƒ
[2] User Failed = CPN set by user, and verification failed.
ƒ
[3] Network Provided = CPN set by network.
Note: Applicable only if Remote Party ID (RPID) header is
enabled.
Send Screening Indicator to
ISDN
[ScreeningInd2ISDN]
Overrides the screening indicator of the calling party's number for
IP-to-Tel ISDN calls.
ƒ
[-1] Not Configured = Not configured (interworking from IP to
ISDN) (default).
ƒ
[0] User Provided = user provided, not screened.
ƒ
[1] User Passed = user provided, verified and passed.
ƒ
[2] User Failed = user provided, verified and failed.
ƒ
[3] Network Provided = network provided.
Add IE in SETUP
[AddIEinSetup]
Adds an optional Information Element (IE) data (in hex format) to
ISDN SETUP messages. For example, to add IE
'0x20,0x02,0x00,0xe1', enter the following value for this
parameter: '200200e1'.
Note: This IE is sent from the Trunk Group IDs defined by the
parameter SendIEonTG.
Trunk Groups to Send IE
[SendIEonTG]
Defines Trunk Group IDs (up to 50 characters) from where the
optional ISDN IE defined by the parameter AddIEinSetup is sent.
For example: '1,2,4,10,12,6'.
Enable User-to-User IE for Tel to
IP
[EnableUUITel2IP]
Enables ISDN PRI-to-SIP interworking.
ƒ
[0] Disable = Disabled (default).
ƒ
[1] Enable = Enable transfer of User-to-User Information
Element (UUIE) from PRI to SIP.
The device supports the following ISDN PRI-to-SIP interworking:
SETUP to SIP INVITE, CONNECT to SIP 200 OK, USER
INFORMATION to SIP INFO, ALERT to SIP 18x response, and
DISCONNECT to SIP BYE response messages.
Note: The interworking of User-to-User IE to SIP INFO is
supported only on the 4ESS PRI variant.
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Parameter
Enable User-to-User IE for IP to
Tel
[EnableUUIIP2Tel]
Description
Enables SIP-to-PRI ISDN interworking.
ƒ
[0] Disable = Disabled (default).
ƒ
[1] Enable = Enable transfer of UUIE from SIP INVITE
message to PRI SETUP message.
The device supports the following SIP-to-PRI ISDN interworking:
SIP INVITE to SETUP, SIP 200 OK to CONNECT, SIP INFO to
USER INFORMATION, SIP 18x to ALERT, and SIP BYE to
DISCONNECT.
Note: The interworking of User-to-User IE to SIP INFO is
supported only on 4ESS PRI variant.
Enable ISDN Tunneling Tel to IP
[EnableISDNTunnelingTel2IP]
Enables ISDN Tunneling.
ƒ
[0] Disable = Disable (default).
ƒ
[1] Using Header = Enable ISDN Tunneling from ISDN PRI to
SIP using a proprietary SIP header.
ƒ
[2] Using Body = Enable ISDN Tunneling from ISDN PRI to
SIP using a dedicated message body.
When ISDN Tunneling is enabled, the device sends all ISDN PRI
messages using the correlated SIP messages. The ISDN SETUP
message is tunneled using SIP INVITE, all mid-call messages
are tunneled using SIP INFO, and ISDN DISCONNECT /
RELEASE is tunneled using SIP BYE messages. The raw data
from the ISDN is inserted into a proprietary SIP header (XISDNTunnelingInfo) or a dedicated message body
(application/isdn) in the SIP messages.
Note: It is necessary to set the parameter
ISDNDuplicateQ931BuffMode to 128 (i.e., duplicate all
messages) for this feature to function.
Enable QSIG Tunneling
[EnableQSIGTunneling]
Enables QSIG tunneling over SIP according to <draft-elwellsipping-qsig-tunnel-03>.
ƒ
[0] Disable = Disable (default).
ƒ
[1] Enable = Enable QSIG tunneling from QSIG to SIP and
vice versa.
When QSIG tunneling is enabled, all QSIG messages are sent as
raw data in corresponding SIP messages using a dedicated
message body.
Notes:
Enable ISDN Tunneling IP to Tel
[EnableISDNTunnelingIP2Tel]
ƒ
QSIG tunneling must be enabled on both originating and
terminating devices.
ƒ
To enable this function, set the parameter
ISDNDuplicateQ931BuffMode to 128 (i.e., duplicate all
messages).
ƒ
[0] Disable = Disable (default).
ƒ
[1] Using Header = Enable ISDN Tunneling from SIP to ISDN
PRI using a proprietary SIP header.
ƒ
[2] Using Body = Enable ISDN Tunneling from SIP to ISDN
PRI using a dedicated message body.
When ISDN Tunneling is enabled, the device extracts raw data
received in a proprietary SIP header (X-ISDNTunnelingInfo) or a
dedicated message body (application/isdn) in the SIP messages
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Parameter
Description
and sends the data as ISDN messages to the PSTN side.
ISDN Transfer On Connect
[SendISDNTransferOnConnect]
Remove CLI when Restricted
[RemoveCLIWhenRestricted]
Remove Calling Name
[RemoveCallingName]
This parameter is used for the ECT/TBCT/RLT/Path
Replacement ISDN Transfer methods. Usually, the device
requests the PBX to connect an incoming and outgoing call. This
parameter determines if the outgoing call (from the device to the
PBX) must be connected before the transfer is initiated.
ƒ
[0] Alert = Enable ISDN Transfer if outgoing call is in Alert or
Connect state (default).
ƒ
[1] Connect = Enable ISDN Transfer only if outgoing call is in
Connect state.
Determines (for IP-to-Tel calls) whether the Calling Number and
Calling Name IEs are removed from the ISDN SETUP message if
the presentation is set to Restricted.
ƒ
[0] No = IE aren't removed (default).
ƒ
[1] Yes = IE are removed.
Enables the device to remove the Calling Name from SIP-toISDN calls.
ƒ
[0] Disable = Does not remove Calling Name (default).
ƒ
[1] Enable = Remove Calling Name.
Default Cause Mapping From
ISDN to SIP
[DefaultCauseMapISDN2IP]
Defines a single default ISDN release cause that is used (in
ISDN-to-IP calls) instead of all received release causes, except
when the following Q.931 cause values are received: Normal Call
Clearing (16), User Busy (17), No User Responding (18), or No
Answer from User (19).
The range is valid Q.931 release causes (0 to 127). The default
value is 0 (i.e., not configured - static mapping is used).
Add Prefix to Redirect Number
[Prefix2RedirectNumber]
Defines a string prefix that is added to the Redirect number
received from the Tel side. This prefix is added to the Redirect
Number in the Diversion header.
The valid range is an 8-character string. The default is an empty
string.
Copy Destination Number to
Redirect Number
[CopyDest2RedirectNumber]
Determines whether the device copies the received ISDN called
number to the outgoing SIP Diversion header for Tel-to-IP calls
(even if a Redirecting Number IE is not received in the ISDN
Setup message). Therefore, the called number is used as a
redirect number. Call redirection information is typically used for
Unified Messaging and voice mail services to identify the
recipient of a message.
Version 5.6
ƒ
[0] Don't copy = Disable (default).
ƒ
[1] Copy after phone number manipulation = Copies the
called number after manipulation. The device first performs
Tel-to-IP destination phone number manipulation (i.e., on the
SIP To header), and only then copies the manipulated called
number to the SIP Diversion header for the Tel-to-IP call.
Therefore, with this option the called and redirected numbers
are identical.
ƒ
[2] Copy before phone number manipulation = Copies the
called number before manipulation. The device first copies the
original called number to the SIP Diversion header, and then
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Parameter
Description
performs Tel-to-IP destination phone number manipulation.
Therefore, this allows you to have different numbers for the
called (i.e., SIP To header) and redirected (i.e., SIP Diversion
header) numbers.
Notes:
Enable Calling Party Category
[EnableCallingPartyCategory]
ƒ
If the incoming ISDN-to-IP call includes a Redirect Number,
this number is overridden by the new called number if this
parameter is set to 1 or 2.
ƒ
This parameter can also be configured for IP Profiles (refer to
''IP Profile Settings'' on page 193).
Determines whether Calling Party Category (CPC) is mapped
between SIP and PRI.
ƒ
[0] Disable = Don't relay the CPC between SIP and PRI
(default).
ƒ
[1] Enable = The CPC is relayed between SIP and PRI.
If enabled, the CPC received in the Originating Line Information
(OLI) IE of an incoming ISDN SETUP message is relayed to the
From / P-Asserted-Identity headers using the 'cpc' parameter, in
the outgoing INVITE message, and vice versa.
For example (calling party is a payphone):
From:<sip:2000;[email protected]>;tag=1c1806157451
Note: This feature is supported only when using the NI-2 PRI
variant.
Digital Out-Of-Service Behavior
[DigitalOOSBehavior]
Determines the method for setting digital trunks to Out-OfService state per device.
ƒ
[0] Default = Uses default behavior for each trunk - see note
below (default)
ƒ
[1] Service = Sends ISDN In or Out of Service (only for ISDN
protocols that support Service message).
ƒ
[2] D-Channel = Takes D-Channel down or up (ISDN only).
ƒ
[3] Alarm = Sends or clears PSTN AIS Alarm (ISDN and
CAS).
ƒ
[4] Block = Blocks trunk (CAS only).
Notes:
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The default behavior (value 0) is as follows:
- ISDN: Use Service messages on supporting variants and
use Alarm on non-supporting variants.
- CAS: Use Alarm.
ƒ
When updating this parameter value at run-time, you must
stop the trunk and then restart it for the update to take effect.
ƒ
To determine the method for setting Out-Of-Service state per
trunk, use the DigitalOOSBehaviorFor Trunk_ID parameter
(refer to ''Trunk Settings'' on page 82).
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Parameter
Description
MLPP (Multilevel Precedence and Preemption)
(Note: For additional MLPP parameters, refer to ''Supplementary Services'' on page 159.)
MLPP Default Namespace
[MLPPDefaultNamespace]
Default Call Priority
[SIPDefaultCallPriority]
Determines the Namespace used for MLPP calls received from
the ISDN side and destined for the Application server. The
Namespace value is not present in the Precedence IE of the PRI
SETUP message. Therefore, the value is used in the ResourcePriority header of the outgoing SIP INVITE request.
ƒ
[1] DSN = DSN (default)
ƒ
[2] DOD = DOD
ƒ
[3] DRSN = DRSN
Defines the default call priority for MLPP calls.
ƒ
[0] 0 = ROUTINE (default)
ƒ
[2] 2 = PRIORITY
ƒ
[6] 6 = IMMEDIATE
ƒ
[8] 8 = FLASH-OVERRIDE
ƒ
[9] 9 = FLASH-OVERRIDE-OVERRIDE
If the incoming SIP INVITE request doesn't contain a valid priority
value in the SIP Resource-Priority header, the default value is
used in the Precedence IE (after translation to the relevant ISDN
Precedence value) of the outgoing PRI SETUP message.
If the incoming PRI SETUP message doesn't contain a valid
Precedence Level value, the default value is used in the
Resource-Priority header of the outgoing SIP INVITE request. In
this scenario, the character string is sent without translation to a
numerical value.
Preemption Tone Duration
[PreemptionToneDuration]
Defines the duration (in seconds) in which the device plays a
preemption tone to both the Tel and IP sides if a call is
preempted.
The valid range is 0 to 60. The default is 3.
Note: If set to 0, no preemption tone is played.
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3.4.8
Advanced Applications
The Advanced Applications menu allows you to configure advanced SIP-based
applications. This menu includes the following page items:
3.4.8.1
„
Voice Mail Settings (refer to ''Configuring the Voice Mail Parameters'' on page 214)
„
RADIUS Parameters (refer to ''Configuring RADIUS Accounting Parameters'' on page
217)
Configuring the Voice Mail (VM) Parameters
The 'Voice Mail Settings' page allows you to configure the voice mail parameters. The voice
mail application applies only to CAS interfaces. For detailed information on voice mail, refer
to the CPE Configuration Guide for Voice Mail User's Manual.
¾ To configure the Voice Mail parameters, take these 4 steps:
1.
Open the 'Voice Mail Settings' page (Configuration tab > Advanced Applications
menu > Voice Mail Settings page item).
Figure 3-87: Voice Mail Settings Page
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2.
Configure the voice mail parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-54: Voice Mail Parameters
Parameter
Description
General
Voice Mail Interface
[VoiceMailInterface]
Enables the voice mail application on the device and
determines the communication method used between the
PBX and the device.
ƒ
[0] None (default)
ƒ
[1] DTMF
ƒ
[2] SMDI
ƒ
[3] QSIG
ƒ
[4] SETUP Only (ISDN)
ƒ
[5] MATRA/AASTRA QSIG
Digit Patterns
The following digit pattern parameters apply only to voice mail applications that use the DTMF
communication method. For the available patterns' syntaxes, refer to the CPE Configuration Guide for
Voice Mail.
Forward on Busy Digit Pattern (Internal)
[DigitPatternForwardOnBusy]
Determines the digit pattern used by the PBX to indicate
'call forward on busy' when the original call is received
from an internal extension.
The valid range is a 120-character string.
Forward on No Answer Digit Pattern
(Internal)
[DigitPatternForwardOnNoAnswer]
Determines the digit pattern used by the PBX to indicate
'call forward on no answer' when the original call is
received from an internal extension.
The valid range is a 120-character string.
Forward on Do Not Disturb Digit Pattern
(Internal)
[DigitPatternForwardOnDND]
Determines the digit pattern used by the PBX to indicate
'call forward on do not disturb' when the original call is
received from an internal extension.
The valid range is a 120-character string.
Forward on No Reason Digit Pattern
(Internal)
[DigitPatternForwardNoReason]
Determines the digit pattern used by the PBX to indicate
'call forward with no reason' when the original call is
received from an internal extension.
The valid range is a 120-character string.
Forward on Busy Digit Pattern (External)
[DigitPatternForwardOnBusyExt]
Determines the digit pattern used by the PBX to indicate
'call forward on busy' when the original call is received
from an external line (not an internal extension).
The valid range is a 120-character string.
Forward on No Answer Digit Pattern
(External)
[DigitPatternForwardOnNoAnswerExt]
Determines the digit pattern used by the PBX to indicate
'call forward on no answer' when the original call is
received from an external line (not an internal extension).
The valid range is a 120-character string.
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Description
Forward on Do Not Disturb Digit Pattern
(External)
[DigitPatternForwardOnDNDExt]
Determines the digit pattern used by the PBX to indicate
'call forward on do not disturb' when the original call is
received from an external line (not an internal extension).
The valid range is a 120-character string.
Forward on No Reason Digit Pattern
(External)
[DigitPatternForwardNoReasonExt]
Determines the digit pattern used by the PBX to indicate
'call forward with no reason' when the original call is
received from an external line (not an internal extension).
The valid range is a 120-character string.
Internal Call Digit Pattern
[DigitPatternInternalCall]
Determines the digit pattern used by the PBX to indicate
an internal call.
The valid range is a 120-character string.
External Call Digit Pattern
[DigitPatternExternalCall]
Determines the digit pattern used by the PBX to indicate
an external call.
The valid range is a 120-character string.
Disconnect Call Digit Pattern
[TelDisconnectCode]
Determines a digit pattern that when received from the
Tel side, indicates the device to disconnect the call.
The valid range is a 25-character string.
Digit To Ignore Digit Pattern
[DigitPatternDigitToIgnore]
A digit pattern that if received as Src (S) or Redirect (R)
numbers is ignored and not added to that number.
The valid range is a 25-character string.
Message Waiting Indication (MWI)
MWI Off Digit Pattern
[MWIOffCode]
Determines the digit code used by the device to notify the
PBX that there aren't any messages waiting for a specific
extension. This code is added as prefix to the dialed
number.
The valid range is a 25-character string.
MWI On Digit Pattern
[MWIOnCode]
Determines the digit code used by the device to notify the
PBX of messages waiting for a specific extension. This
code is added as prefix to the dialed number.
The valid range is a 25-character string.
MWI Suffix Pattern
[MWISuffixCode]
Determines the digit code used by the device as a suffix
for 'MWI On Digit Pattern' and 'MWI Off Digit Pattern'.
This suffix is added to the generated DTMF string after
the extension number.
The valid range is a 25-character string.
MWI Source Number
[MWISourceNumber]
Determines the calling party's phone number used in the
Q.931 MWI SETUP message to PSTN. If not configured,
the channel's phone number is used as the calling
number.
SMDI
Enable SMDI
[SMDI]
Enables Simplified Message Desk Interface (SMDI)
interface on the device.
ƒ
[0] Disable = Normal serial (default).
ƒ
[1] Enable (Bellcore)
ƒ
[2] Ericsson MD-110
ƒ
[3] NEC (ICS)
Note: When the RS-232 connection is used for SMDI
messages (Serial SMDI), it cannot be used for other
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Parameter
Description
applications, for example, to access the Command Line
Interface (CLI).
SMDI Timeout
[SMDITimeOut]
3.4.8.2
Determines the time (in msec) that the device waits for an
SMDI Call Status message before or after a SETUP
message is received. This parameter synchronizes the
SMDI and analog CAS interfaces.
If the timeout expires and only an SMDI message is
received, the SMDI message is dropped. If the timeout
expires and only a SETUP message is received, the call
is established.
The valid range is 0 to 10000 (i.e., 10 seconds). The
default value is 2000.
Configuring RADIUS Accounting Parameters
The 'RADIUS Parameters' page is used for configuring the Remote Authentication Dial In
User Service (RADIUS) accounting parameters.
¾ To configure the RADIUS parameters, take these 4 steps:
1.
Open the ‘RADIUS Parameters' page (Configuration tab > Advanced Applications
menu > RADIUS Parameters page item).
Figure 3-88: RADIUS Parameters Page
2.
Configure the RADIUS accounting parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-55: RADIUS Parameters Description
Parameter
Enable RADIUS Access
Control
EnableRADIUS
Description
Enables or disables the RADIUS application.
ƒ
[0] Disable = disables RADIUS application (default)
ƒ
[1] Enable = enables RADIUS application
Accounting Server IP
Address
[RADIUSAccServerIP]
IP address of the RADIUS accounting server.
Accounting Port
[RADIUSAccPort]
Port of the RADIUS accounting server.
The default value is 1646.
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Parameter
Description
RADIUS Accounting Type
[RADIUSAccountingType]
AAA Indications
[AAAIndications]
3.4.9
Determines when the RADIUS accounting messages are sent to the
RADIUS accounting server.
ƒ
[0] At Call Release = Sent at call release only (default).
ƒ
[1] At Connect & Release = Sent at call connect and release.
ƒ
[2] At Setup & Release = Sent at call setup and release.
Determines the Authentication, Authorization and Accounting (AAA)
indications.
ƒ
[0] None = No indications (default).
ƒ
[3] Accounting Only = Only accounting indications are used.
Configuring the TDM Bus Settings
The device's Time-Division Multiplexing (TDM) bus settings can be performed in the 'TDM
Bus Settings' page, as described in the procedure below.
¾ To configure the TDM Bus settings, take these 5 steps:
1.
Open the 'TDM Bus Settings' page (Configuration tab > TDM Configuration menu >
TDM Bus Settings page item).
Figure 3-89: TDM Bus Settings Page
2.
Configure the TDM bus parameters according to the table below.
3.
Click the Submit button to save your changes.
4.
Save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
5.
Reset the device (refer to ''Resetting the Device'' on page 228).
Table 3-56: TDM Bus Settings Parameters Description
Parameter
PCM Law Select
[PCMLawSelect]
Description
Determines the type of PCM companding law in input/output TDM bus.
ƒ
[1] Alaw = Alaw (default)
ƒ
[3] MuLaw = MuLaw
Note: Typically, A-Law is used for E1 spans and μ-Law for T1/J1 spans.
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Parameter
Description
Idle PCM Pattern
[IdlePCMPattern]
Defines the PCM Pattern that is applied to the E1/T1 timeslot (Bchannel) when the channel is idle.
The range is 0 to 255. The default is set internally according to the Law
select 1 (0xFF for Mu-Law; 0x55 for A-law).
Idle ABCD Pattern
[IdleABCDPattern]
Defines the ABCD (CAS) Pattern that is applied to the CAS signaling
bus when the channel is idle.
The valid range is 0x0 to 0xF. The default is -1 (i.e., default pattern =
0000).
Note: This parameter is applicable only when using PSTN interface with
CAS protocols.
TDM Bus Local Reference
[TDMBusLocalReference
]
Physical Trunk ID from which the device recovers (receives) its clock
synchronization.
The range is 0 to the maximum number of Trunks. The default is Trunk
ID 1.
Note: This parameter is applicable only if the parameter
TDMBusClockSource is set to 4 and the parameter
TDMBusPSTNAutoClockEnable is set to 0.
TDM Bus PSTN Auto
Clock
[TDMBusPSTNAutoCloc
kEnable]
Enables or disables the PSTN trunk Auto-Fallback Clock feature.
ƒ
[0] Disable (default) = Recovers the clock from the E1/T1 line
defined by the parameter TDMBusLocalReference.
ƒ
[1] Enable = Recovers the clock from any connected synchronized
slave E1/T1 line. If this trunk loses its synchronization, the device
attempts to recover the clock from the next trunk. Note that initially,
the device attempts to recover the clock from the trunk defined by
the parameter TDMBusLocalReference.
Note: This parameter is relevant only if the parameter
TDMBusClockSource is set to 4.
TDM Bus PSTN Auto
Clock Reverting
[TDMBusPSTNAutoCloc
kRevertingEnable]
Enables or disables the PSTN trunk auto-fallback reverting feature. If
enabled and a trunk returning to service has an AutoClockTrunkPriority
parameter value (refer to ''Configuring the Trunk Settings'' on page 82)
that is higher than the priority of the local reference trunk (set in the
TDMBusLocalReference parameter), the local reference reverts to the
trunk with the higher priority that has returned to service for the device's
clock source.
ƒ
[0] Disable (default)
ƒ
[1] Enable
Note: This parameter is applicable only when the
TDMBusPSTNAutoClockEnable parameter is set to 1.
TDM Bus Clock Source
[TDMBusClockSource]
Selects the clock source to which the device synchronizes.
ƒ
[1] Internal = Generate clock from local source (default).
ƒ
[4] Network = Recover clock from PSTN line.
For detailed information on configuring the device's clock settings, refer
to ''Clock Settings'' on page 393.
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3.5
Management Tab
The Management tab on the Navigation bar displays all menus related to device
management. These menus appear in the Navigation tree and include the following:
3.5.1
„
Management Configuration (refer to ''Management Configuration'' on page 220)
„
Software Update (refer to ''Software Update'' on page 231)
Management Configuration
The Management Configuration menu allows you to configure the device's management
parameters. This menu contains the following page items:
3.5.1.1
„
Management Settings (refer to ''Configuring the Management Settings'' on page 220)
„
Regional settings (refer to ''Configuring the Regional Settings'' on page 227)
„
Maintenance Actions (refer to ''Maintenance Actions'' on page 228)
Configuring the Management Settings
The 'Management Settings' page allows you to configure the device's management
parameters.
¾ To configure the Management parameters, take these 4 steps:
1.
Open the 'Management Settings' page (Management
Configuration menu > Management Settings page item).
tab >
Management
Figure 3-90: Management Settings Page
2.
Configure the Management Settings according to the table below.
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3.
Click the Submit button to save your changes.
4.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Table 3-57: Management Settings Parameters
Parameter
Description
Syslog Settings
Syslog Server IP Address
[SyslogServerIP]
IP address (in dotted-decimal notation) of the computer you are
using to run the Syslog server. The Syslog server is an
application designed to collect the logs and error messages
generated by the device.
Default IP address is 0.0.0.0.
For information on Syslog, refer to the Product Reference
Manual.
Syslog Server Port
[SyslogServerPort]
Defines the UDP port of the Syslog server.
The valid range is 0 to 65,535. The default port is 514.
For information on the Syslog, refer to the Product Reference
Manual.
Enable Syslog
[EnableSyslog]
Sends the logs and error message generated by the device to the
Syslog server.
ƒ
[0] Disable = Logs and errors are not sent to the Syslog server
(default).
ƒ
[1] Enable = Enables the Syslog server.
Notes:
ƒ
If you enable Syslog, you must enter an IP address and a port
number (using SyslogServerIP and SyslogServerPort
parameters).
ƒ
You can configure the device to send Syslog messages
implementing Debug Recording (refer to Debug Recording
(DR)), by using the SyslogOutputMethod ini file parameter.
ƒ
Syslog messages may increase the network traffic.
ƒ
To configure Syslog logging levels, use the parameter
GwDebugLevel, as described in ''Advanced Parameters'' on
page 151.
ƒ
For information on the Syslog, refer to the Product Reference
Manual.
SNMP Settings
For detailed information on the SNMP parameters that can be configured via the ini file, refer to
''SNMP Parameters'' on page 282. For detailed information on developing an SNMP-based program to
manage your device, refer to the Product Reference Manual.
SNMP Trap Destinations
button to configure the SNMP trap
Click the arrow
destinations (refer to ''Configuring the SNMP Trap Destinations
Table'' on page 222).
SNMP Community String
button to configure the SNMP community
Click the arrow
strings (refer to ''Configuring the SNMP Community Strings'' on
page 224).
SNMP V3 Table
button to configure the SNMP V3 users (refer
Click the arrow
to ''Configuring SNMP V3 Table'' on page 225).
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Parameter
Description
SNMP Trusted Managers
button to configure the SNMP Trusted
Click the arrow
Managers (refer to ''Configuring SNMP Trusted Managers'' on
page 226).
Enable SNMP
[DisableSNMP]
ƒ
[0] Enable = SNMP is enabled (default).
ƒ
[1] Disable = SNMP is disabled and no traps are sent.
Trap Manager Host Name
[SNMPTrapManagerHostName]
Defines an FQDN of a remote host that is used as an SNMP
manager. The resolved IP address replaces the last entry in the
Trap Manager table (defined by the parameter
SNMPManagerTableIP_x) and the last trap manager entry of
snmpTargetAddrTable in the snmpTargetMIB.
For example: 'mngr.corp.mycompany.com'.
The valid range is a 99-character string.
Activity Types to Report via 'Activity Log' Messages
The Activity Log mechanism enables the device to send log messages (to a Syslog server) for
reporting on certain types of Web operations according to the below user-defined filters.
Parameters Value Change
[ActivityListToLog = PVC]
Changes made on-the-fly to parameters.
Auxiliary Files Loading
[ActivityListToLog = AFL]
Loading of auxiliary files (e.g., via 'Certificate' page).
Device Reset
[ActivityListToLog = DR]
Reset of device via the 'Maintenance Actions' page.
Flash Memory Burning
[ActivityListToLog = FB]
Burning of files / parameters to flash (e.g., 'Maintenance Actions'
page).
Device Software Update
[ActivityListToLog = SWU]
cmp loading via the Software Upgrade Wizard.
Access to Restricted Domains
[ActivityListToLog = ARD]
Access to Restricted Domains, which includes the following
pages:
ƒ
ini parameters (AdminPage)
ƒ
General Security Settings
ƒ
Configuration File
ƒ
IPSec/IKE tables
ƒ
Software Upgrade Key
ƒ
Internal Firewall
ƒ
Web Access List
ƒ
Web User Accounts
Non-Authorized Access
[ActivityListToLog = NAA]
Attempt to access the Web interface with a false / empty user
name or password.
Sensitive Parameters Value
Change
[ActivityListToLog = SPC]
Changes made to sensitive parameters:
(1) IP Address
(2) Subnet Mask
(3) Default Gateway IP Address
(4) ActivityListToLog
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3.5.1.1.1 Configuring the SNMP Trap Destinations Table
The 'SNMP Trap Destinations' page allows you to configure up to five SNMP trap
managers.
¾ To configure the SNMP Trap Destinations table, take these 5 steps:
1.
Access the 'Management Settings' page, as described in ''Configuring the
Management Settings'' on page 220.
2.
In the 'SNMP Trap Destinations' field, click the right-pointing arrow
'SNMP Trap Destinations' page appears.
button; the
Figure 3-91: SNMP Trap Destinations Page
3.
Configure the SNMP trap managers parameters according to the table below.
4.
Click the Submit button to save your changes.
5.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Note: Only table row entries whose corresponding check boxes are selected are
applied when clicking Submit; otherwise, settings revert to their defaults.
Table 3-58: SNMP Trap Destinations Parameters Description
Parameter
Description
SNMP Manager
[SNMPManagerIsUsed_x]
Determines the validity of the parameters (IP address and port
number) of the corresponding SNMP Manager used to receive SNMP
traps.
ƒ
[0] (Check box cleared) = Disabled (default)
ƒ
[1] (Check box selected) = Enabled
IP Address
[SNMPManagerTableIP_x]
IP address of the remote host used as an SNMP Manager. The device
sends SNMP traps to these IP addresses.
Enter the IP address in dotted-decimal notation, e.g., 108.10.1.255.
Trap Port
[SNMPManagerTrapPort_x
]
Defines the port number of the remote SNMP Manager. The device
sends SNMP traps to these ports.
The valid SNMP trap port range is 100 to 4000. The default port is
162.
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Parameter
Description
Trap Enable
[SNMPManagerTrapSendi
ngEnable_x]
Activates or de-activates the sending of traps to the corresponding
SNMP Manager.
ƒ
[0] Disable = Sending is disabled.
ƒ
[1] Enable = Sending is enabled (default).
3.5.1.1.2 Configuring the SNMP Community Strings
The 'SNMP Community String' page allows you to configure up to five read-only and up to
five read-write SNMP community strings, and to configure the community string that is used
for sending traps. For detailed information on SNMP community strings, refer to the Product
Reference Manual.
¾ To configure the SNMP community strings, take these 5 steps:
1.
Access the 'Management Settings' page, as described in ''Configuring the
Management Settings'' on page 220.
2.
In the 'SNMP Community String' field, click the right-pointing arrow
'SNMP Community String' page appears.
button; the
Figure 3-92: SNMP Community Strings Page
3.
Configure the SNMP community strings parameters according to the table below.
4.
Click the Submit button to save your changes.
5.
To save the changes to flash memory, refer to ''Saving Configuration'' on page 230.
Note: To delete a community string, select the Delete check box corresponding to
the community string that you want to delete, and then click Submit.
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Table 3-59: SNMP Community Strings Parameters Description
Parameter
Description
Community String
Trap Community String
[SNMPTrapCommunityString]
ƒ
Read Only [SNMPReadOnlyCommunityString_x]: Up to five
read-only community strings (up to 19 characters each). The
default string is 'public'.
ƒ
Read / Write [SNMPReadWriteCommunityString_x]: Up to
five read / write community strings (up to 19 characters each).
The default string is 'private'.
Community string used in traps (up to 19 characters).
The default string is 'trapuser'.
3.5.1.1.3 Configuring SNMP V3 Users
The 'SNMP V3 Settings' page allows you to configure authentication and privacy for up to
10 SNMP v3 users.
¾ To configure the SNMP v3 users, take the following 6 steps:
1.
Access the 'Management Settings' page, as described in ''Configuring the
Management Settings'' on page 220.
2.
In the 'SNMP V3 Table' field, click the right-pointing arrow
Settings' page appears.
button; the 'SNMP V3
Figure 3-93: SNMP V3 Setting Page
3.
To add an SNMP v3 user, in the 'Add' field, enter the desired row index, and then click
Add. A new row appears.
4.
Configure the SNMP V3 Setting parameters according to the table below.
5.
Click the Apply button to save your changes.
6.
To save the changes, refer to ''Saving Configuration'' on page 230.
Notes:
Version 5.6
•
For a description of the web interface's table command buttons (e.g.,
Duplicate and Delete), refer to ''Working with Tables'' on page 30.
•
You can also configure SNMP v3 users using the ini file table parameter
SNMPUsers (refer to ''SNMP Parameters'' on page 282).
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Table 3-60: SNMP V3 Users Parameters
Parameter
Description
Index
[SNMPUsers_Index]
The table index.
The valid range is 0 to 9.
User Name
[SNMPUsers_Username]
Name of the SNMP v3 user. This name must be unique.
Authentication Protocol
[SNMPUsers_AuthProtocol]
Authentication protocol of the SNMP v3 user.
Privacy Protocol
[SNMPUsers_PrivProtocol]
ƒ
[0] None (default)
ƒ
[1] MD5
ƒ
[2] SHA-1
Privacy protocol of the SNMP v3 user.
ƒ
[0] None (default)
ƒ
[1] DES
ƒ
[2] 3DES
ƒ
[3] AES-128
ƒ
[4] AES-192
ƒ
[5] AES-256
Authentication Key
[SNMPUsers_AuthKey]
Authentication key. Keys can be entered in the form of a text
password or long hex string. Keys are always persisted as long hex
strings and keys are localized.
Privacy Key
[SNMPUsers_PrivKey]
Privacy key. Keys can be entered in the form of a text password or
long hex string. Keys are always persisted as long hex strings and
keys are localized.
Group
[SNMPUsers_Group]
The group with which the SNMP v3 user is associated.
ƒ
[0] Read-Only (default)
ƒ
[1] Read-Write
ƒ
[2] Trap
Note: All groups can be used to send traps.
3.5.1.1.4 Configuring SNMP Trusted Managers
The 'SNMP Trusted Managers' page allows you to configure up to five SNMP Trusted
Managers, based on IP addresses. By default, the SNMP agent accepts SNMP Get and
Set requests from any IP address, as long as the correct community string is used in the
request. Security can be enhanced by using Trusted Managers, which is an IP address
from which the SNMP agent accepts and processes SNMP requests.
¾ To configure the SNMP Trusted Managers, take the following 6
steps:
1.
Access the 'Management Settings' page, as described in ''Configuring the
Management Settings'' on page 220.
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In the 'SNMP Trusted Managers' field, click the right-pointing arrow
'SNMP Trusted Managers' page appears.
button; the
Figure 3-94: SNMP Trusted Managers
3.5.1.2
3.
Select the check box corresponding to the SNMP Trusted Manager that you want to
enable and for whom you want to define an IP address.
4.
Define an IP address in dotted-decimal notation.
5.
Click the Submit button to apply your changes.
6.
To save the changes, refer to ''Saving Configuration'' on page 230.
Configuring the Regional Settings
The 'Regional Settings' page allows you to define and view the device's internal date and
time.
¾ To configure the device's date and time, take these 3 steps:
1.
Open the 'Regional Settings' page (Management tab > Management Configuration
menu > Regional Settings page item).
Figure 3-95: Regional Settings Page
2.
Enter the current date and time in the geographical location in which the device is
installed.
3.
Click the Submit button; the date and time are automatically updated.
Notes:
Version 5.6
•
If the device is configured to obtain the date and time from an SNTP
server (refer to ''Configuring the Application Settings'' on page 57), the
fields on this page are read-only and cannot be modified. For an
explanation on SNTP, refer to ''Simple Network Time Protocol Support''
on page 383.
•
After performing a hardware reset, the date and time are returned to their
defaults and therefore, should be updated.
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3.5.1.3
Maintenance Actions
The 'Maintenance Actions' page allows you to perform the following operations:
„
Reset the device (refer to ''Resetting the Device'' on page 228)
„
Lock and unlock the device (refer to ''Locking and Unlocking the Device'' on page 229)
„
Save the configuration to the device's flash memory (refer to ''Saving Configuration'' on
page 230)
¾ To access the 'Maintenance Actions' page, take this step:
„
On the Navigation bar, click the Management tab, and then in the Navigation tree,
select the Management Configuration menu, and then choose the Maintenance
Actions page item.
Figure 3-96: Maintenance Actions Page
3.5.1.3.1 Resetting the Device
The 'Maintenance Actions' page allows you to remotely reset the device. In addition, before
resetting the device, you can choose the following options:
„
Save the device's current configuration to the device's flash memory (non-volatile).
„
Perform a graceful shutdown, i.e., device reset starts only after a user-defined time
expires (i.e., timeout) or after no more active traffic exists (the earliest thereof).
¾ To reset the device, take these 6 steps:
1.
Open the 'Maintenance Actions' page (refer to ''Maintenance Actions'' on page 228).
2.
Under the 'Reset Configuration' group, from the 'Burn To FLASH' drop-down list, select
one of the following options:
•
'Yes': The device's current configuration is saved (burned) to the flash memory
prior to reset (default).
•
'No': Resets the device without saving the current configuration to flash (discards
all unsaved modifications).
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Under the 'Reset Configuration' group, from the 'Graceful Option' drop-down list, select
one of the following options:
•
'Yes': Reset starts only after the user-defined time in the 'Shutdown Timeout' field
(refer to Step 4) expires or after no more active traffic exists (the earliest thereof).
In addition, no new traffic is accepted.
•
'No': Reset starts regardless of traffic, and any existing traffic is terminated at
once.
4.
In the 'Shutdown Timeout' field (relevant only if the 'Graceful Option' in the previous
step is set to 'Yes'), enter the time after which the device resets. Note that if no traffic
exists and the time has not yet expired, the device resets.
5.
Click the Reset button; a confirmation message box appears, requesting you to
confirm.
Figure 3-97: Reset Confirmation Message Box
6.
Click OK to confirm device reset; if the parameter 'Graceful Option' is set to 'Yes' (in
Step 3), the reset is delayed and a screen displaying the number of remaining calls
and time is displayed. When the device begins to reset, a message appears notifying
you of this.
Notes:
•
Throughout the Web interface, parameters preceded by the lightning
symbol are not applied on-the-fly to the device and require that you reset
the device for them to take effect.
•
If you modify parameters that only take effect after a device reset, after
you click the Submit button, the toolbar displays the word 'Reset' (refer to
''Toolbar'' on page 21) to remind you to later reset the device.
3.5.1.3.2 Locking and Unlocking the Device
The Lock and Unlock options allow you to lock the device so that it doesn't accept any new
incoming calls. This is useful when, for example, you are uploading new software files to
the device and you don't want any traffic to interfere with the process.
¾ To lock the device, take these 5 steps:
1.
Open the 'Maintenance Actions' page (refer to ''Maintenance Actions'' on page 228).
2.
Under the 'LOCK / UNLOCK' group, from the 'Graceful Option' drop-down list, select
one of the following options:
•
Version 5.6
'Yes': The device is 'locked' only after the user-defined time in the 'Lock Timeout'
field (refer to Step 3) expires or no more active traffic exists (the earliest thereof).
In addition, no new traffic is accepted.
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•
'No': The device is 'locked' regardless of traffic. Any existing traffic is terminated
immediately.
Note: These options are only available if the current status of the device is in the
Unlock state.
3.
In the 'Lock Timeout' field (relevant only if the parameter 'Graceful Option' in the
previous step is set to 'Yes'), enter the time (in seconds) after which the device locks.
Note that if no traffic exists and the time has not yet expired, the device locks.
4.
Click the LOCK button; a confirmation message box appears requesting you to
confirm device Lock.
Figure 3-98: Device Lock Confirmation Message Box
5.
Click OK to confirm device Lock; if 'Graceful Option' is set to 'Yes', the lock is delayed
and a screen displaying the number of remaining calls and time is displayed.
Otherwise, the lock process begins immediately. The 'Current Admin State' field
displays the current state: LOCKED or UNLOCKED.
¾ To unlock the device, take these 2 steps:
1.
Open the 'Maintenance Actions' page (refer to ''Maintenance Actions'' on page 228).
2.
Under the 'LOCK / UNLOCK' group, click the UNLOCK button. Unlock starts
immediately and the device accepts new incoming calls.
3.5.1.3.3 Saving Configuration
The 'Maintenance Actions' page allows you to save (burn) the current parameter
configuration (including loaded auxiliary files) to the device's non-volatile memory (i.e.,
flash). The parameter modifications that you make throughout the Web interface's pages
are temporarily saved (to the volatile memory - RAM) when you click the Submit button on
these pages. Parameter settings that are only saved to the device's RAM revert to their
previous settings after a hardware/software reset (or power failure). Therefore, to ensure
that your configuration changes are retained, you must save them to the device's flash
memory using the burn option described below.
¾ To save the changes to the non-volatile flash memory , take these
2 steps:
1.
Open the 'Maintenance Actions' page (refer to ''Maintenance Actions'' on page 228).
2.
Under the 'Save Configuration' group, click the BURN button; a confirmation message
appears when the configuration successfully saves.
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Notes:
3.5.2
•
Saving configuration to the non-volatile memory may disrupt traffic on the
device. To avoid this, disable all new traffic before saving, by performing
a graceful lock (refer to ''Locking and Unlocking the Device'' on page
229).
•
Throughout the Web interface, parameters preceded by the lightning
symbol are not applied on-the-fly to the device and require that you reset
the device (refer to ''Resetting the Device'' on page 228) for them to take
effect.
Software Update
The Software Update menu allows you to upgrade the device's software by loading a new
cmp file (compressed firmware) along with the ini file and a suite of auxiliary files, or to
update existing auxiliary files.
The Software Update menu includes the following page items:
3.5.2.1
„
Load Auxiliary Files (refer to ''Loading Auxiliary Files'' on page 231)
„
Software Update Key (refer to “Upgrading the Software Upgrade Key” on page 233)
„
Software Upgrade Wizard (refer to ''Software Upgrade Wizard'' on page 236)
„
Configuration File (refer to ''Backing Up and Restoring Configuration'' on page 240)
Loading Auxiliary Files
The 'Load Auxiliary Files' page allows you to load to the device various auxiliary files
(described in the table below). For detailed information on these files, refer to ''Auxiliary
Configuration Files'' on page 335. For information on deleting these files from the device,
refer to ''Device Information'' on page 244.
Table 3-61: Auxiliary Files Descriptions
File Type
ini
Description
Provisions the device’s parameters. The Web interface enables practically full
device provisioning, but customers may occasionally require new feature
configuration parameters in which case this file is loaded.
Note: Loading this file only provisions those parameters that are included in the
ini file. Parameters that are not specified in the ini file are reset to factory default
values.
CAS
Up to eight different CAS files containing specific CAS protocol definitions for
digital modules. These files are provided to support various types of CAS
signaling.
Voice Prompts
The voice announcement file contains a set of Voice Prompts (VP) that are
played by the device during operation.
Dial Plan
Dial plan file.
Call Progress
Tones
This is a region-specific, telephone exchange-dependent file that contains the
Call Progress Tones (CPT) levels and frequencies that the device uses. The
default CPT file is: U.S.A.
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File Type
Description
Prerecorded
Tones
The dat PRT file enhances the device's capabilities of playing a wide range of
telephone exchange tones that cannot be defined in the Call Progress Tones file.
User Info
The User Information file maps PBX extensions to IP numbers. This file can be
used to represent PBX extensions as IP phones in the global 'IP world'.
¾ To load an auxiliary file to the device using the Web interface, take
these 6 steps:
1.
Open the 'Load Auxiliary Files' page (Management tab > Software Update menu >
Load Auxiliary Files page item).
Figure 3-99: Load Auxiliary Files Page
2.
Click the Browse button corresponding to the file type that you want to load, navigate
to the folder in which the file is located, and then click Open; the name and path of the
file appear in the field next to the Browse button.
3.
Click the Load File button corresponding to the file you want to load.
4.
Repeat steps 2 through 3 for each file you want to load.
5.
To save the loaded auxiliary files to flash memory, refer to ''Saving Configuration'' on
page 230.
6.
To reset the device (if you have loaded a Call Progress Tones file), refer to ''Resetting
the Device'' on page 228.
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Notes:
•
Saving an auxiliary file to flash memory may disrupt traffic on the device.
To avoid this, disable all traffic on the device by performing a graceful
lock (refer to ''Locking and Unlocking the Device'' on page 229).
•
You can schedule automatic loading of updated auxiliary files using
HTTP, HTTPS, FTP, or NFS (refer to the Product Reference Manual).
You can also load the Auxiliary files using the ini file. Before you load the files to the device,
in the ini file you need to include certain ini file parameters associated with these files.
These ini file parameters specify the files that you want loaded and whether they must be
stored in the non-volatile memory. For a description of the ini file parameters associated
with the auxiliary files, refer to ''Configuration Files Parameters'' on page 331.
¾ To load the auxiliary files via the ini file, take these 3 steps:
3.5.2.2
1.
In the ini file, define the auxiliary files to be loaded to the device. You can also define in
the ini file whether the loaded files must be stored in the non-volatile memory so that
the TFTP process is not required every time the device boots up.
2.
Save the auxiliary files you want to load and the ini file in the same directory on your
PC.
3.
Invoke a BootP/TFTP session; the ini and auxiliary files are loaded to the device.
Upgrading the Software Upgrade Key
The device is supplied with a Software Upgrade Key for each of its TrunkPack Modules
(TPM). You can upgrade the device's features, capabilities, and quantity of available
resources by by purchasing a new key to match your requirements. The Software Upgrade
Key is provided in string format in a text file, which is loaded to the device's non-volatile
flash memory. The string defines the device's allowed features and capabilities. A new key
overwrites a previously installed key.
You can load a Software Upgrade Key using one of the following:
„
Web interface
„
BootP/TFTP configuration utility (refer to ''Loading via BootP/TFTP'' on page 235)
„
AudioCodes’ EMS (refer to AudioCodes’ EMS User’s Manual or EMS Product
Description)
Warning: Don't modify the contents of the Software Upgrade Key file.
Notes:
•
The Software Upgrade Key is an encrypted key. Each TPM utilizes a
unique key.
•
The Software Upgrade Key is provided only by AudioCodes.
The procedure below describes how to load a Software Upgrade Key to the device using
the Web interface.
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¾ To load a Software Upgrade Key, take these 6 steps:
1.
Open the 'Software Upgrade Key Status' page (Management tab > Software Update
menu > Software Upgrade Key page item).
Figure 3-100: Software Upgrade Key Status Page
2.
3.
Backup your current Software Upgrade Key as a precaution so that you can re-load
this backup key to restore the device's original capabilities if the new key doesn’t
comply with your requirements:
a.
In the 'Current Key' field, copy the string of text and paste it in any standard text
file.
b.
Save the text file to a folder on your PC with a name of your choosing.
Open the new Software Upgrade Key file and ensure that the first line displays
'[LicenseKeys]' and that it contains one or more lines in the following format:
S/N<serial number of the first or second module> = <long Software Upgrade Key>
For example: S/N370604 = jCx6r5tovCIKaBBbhPtT53Yj...
One S/N must match the serial number of your device. The device’s serial number can
be viewed in the ‘Device Information’ page (refer to ''Device Information'' on page 244).
4.
Follow one of the following procedures, depending on whether you are loading a single
or multiple key S/N lines:
•
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Single key S/N line:
a.
Open the Software Upgrade Key text file (using, for example, Microsoft®
Notepad).
b.
Select and copy the key string of the device's S/N and paste it into the field
'Add a Software Upgrade Key'.
c.
Click the Add Key button.
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Multiple S/N lines (as shown below):
Figure 3-101: Software Upgrade Key with Multiple S/N Lines
5.
6.
a.
in the 'Send Upgrade Key file' field, click the Browse button and navigate to
the folder in which the Software Upgrade Key text file is located on your PC.
b.
Click the Send File button; the new key is loaded to the device and
validated. If the key is valid, it is burned to memory and displayed in the
'Current Key' field.
Verify that the Software Upgrade Key file was successfully loaded to the device, by
using one of the following methods:
•
In the ‘Key features’ group, ensure that the features and capabilities activated by
the installed string match those that were ordered.
•
Access the Syslog server (refer to the Product Reference Manual) and ensure
that the following message appears in the Syslog server:
"S/N___ Key Was Updated. The Board Needs to be Reloaded with ini file\n".
Reset the device; the new capabilities and resources are active.
Note: If the Syslog server indicates that the Software Upgrade Key file was
unsuccessfully loaded (i.e., the 'SN_' line is blank), perform the following
preliminary troubleshooting procedures:
1.
Open the Software Upgrade Key file and check that the S/N line
appears. If it does not appear, contact AudioCodes.
2.
Verify that you’ve loaded the correct file. Open the file and ensure that
the first line displays [LicenseKeys].
3.
Verify that the contents of the file has not been altered in any way.
3.5.2.2.1 Loading via BootP/TFTP
The procedure below describes how to load a Software Upgrade Key to the device using
AudioCodes' BootP/TFTP Server utility (for a detailed description on the BootP utility, refer
to the Product Reference Manual).
¾ To load a Software Upgrade Key file using BootP/TFTP, take these
6 steps:
1.
Place the Software Upgrade Key file (typically, a *.txt file) in the same folder in which
the device's cmp file is located.
2.
Start the BootP/TFTP Server utility.
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3.
From the Services menu, choose Clients; the 'Client Configuration' screen is
displayed.
4.
From the 'INI File' drop-down list, select the Software Upgrade Key file. Note that the
device's cmp file must be specified in the 'Boot File' field.
5.
Configure the initial BootP/TFTP parameters as required, and then click OK.
6.
Reset the device; the cmp and Software Upgrade Key files are loaded to the device.
Note: To load the Software Upgrade Key using BootP/TFTP, the extension name of
the key file must be *.ini.
3.5.2.3
Software Upgrade Wizard
The Software Upgrade Wizard guides you through the process of software upgrade:
selecting files and loading them to the device. The wizard also enables you to upgrade
software while maintaining the existing configuration. Using the wizard obligates you to load
and burn a cmp file to the device. You can choose to also use the wizard to load the ini and
auxiliary files (e.g., Call Progress Tones), but this option cannot be pursued without loading
the cmp file. For the ini and each auxiliary file type, you can choose to reload an existing
file, load a new file, or not load a file at all.
The Software Upgrade Wizard allows you to load the following files:
„
cmp (mandatory) - compressed firmware file
„
ini - configuration file
„
Auxiliary files: CPT (Call Progress Tone), VP (Voice Prompts), PRT (Prerecorded
Tones), CAS, and USRINF (User Info)
Warnings:
•
Before upgrading the device to a new major software version (e.g., from
version 5.4 to 5.6), save a copy of the device's configuration settings (i.e.,
ini file) to your PC (refer to ''Backing Up and Restoring Configuration'' on
page 240) and ensure that you have all the original auxiliary files (e.g.,
CPT file) currently being used by the device. After you have upgraded the
device, upload these files to the device.
•
The Software Upgrade Wizard requires the device to be reset at the end
of the process, which may disrupt its traffic. To avoid this, disable all
traffic on the device before initiating the wizard by performing a graceful
lock (refer to ''Locking and Unlocking the Device'' on page 229).
•
Before you can load an ini or any auxiliary file, you must first load a cmp
file.
•
When you activate the wizard, the rest of the Web interface is
unavailable. After you load the desired files, access to the full Web
interface is restored.
•
You can schedule automatic loading of cmp, ini, and auxiliary files using
HTTP, HTTPS, FTP, or NFS. (Refer to the Product Reference Manual).
Notes:
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¾ To use the Software Upgrade Wizard, take these 11 steps:
1.
Stop all traffic on the device (refer to the note above).
2.
Open the 'Software Upgrade Wizard' (Management tab > Software Update menu >
Software Upgrade Wizard); the 'Software Upgrade Wizard' page appears.
Figure 3-102: Start Software Upgrade Wizard Screen
3.
Click the Start Software Upgrade button; the 'Load a CMP file' Wizard page appears.
Figure 3-103: Load CMP File Wizard Page
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Note: At this stage, you can quit the Software Update Wizard, by clicking Cancel
, without requiring a device reset. However, once you start uploading a
cmp file, the process must be completed with a device reset.
4.
Click the Browse button, navigate to the cmp file, and then click Send File; the cmp
file is loaded to the device and you're notified as to a successful loading, as shown
below.
Figure 3-104: CMP File Successfully Loaded Message
5.
Click one of the following buttons:
•
•
Reset; the device resets with the newly loaded cmp, and utilizing the current
configuration and auxiliary files.
Next; the 'Load an ini File' wizard page opens.
Note that as you progress by clicking Next, the relevant file name corresponding to the
applicable Wizard page is highlighted in the file list on the left.
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In the 'Load an ini File' page, you can now choose to either:
•
Click Browse, navigate to the ini file, and then click Send File; the ini file is
loaded to the device and you're notified as to a successful loading.
•
Use the ini file currently used by the device, by not selecting an ini file and by
ensuring that the 'Use existing configuration' check box is marked (default).
•
Return the device's configuration settings to factory defaults, by not selecting an
ini file and by clearing the 'Use existing configuration' check box.
Figure 3-105: Load an ini File Wizard Page
7.
8.
Version 5.6
You can now choose to either:
•
Click Reset; the device resets, utilizing the new cmp and ini file you loaded up to
now as well as utilizing the other auxiliary files.
•
Click Back; the 'Load a cmp file' page is opened again.
•
Click Next; the next page opens for loading the next consecutive auxiliary file
listed in the Wizard.
Follow the same procedure as for loading the ini file (Step 6) for loading the auxiliary
files.
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9.
In the 'FINISH' page, complete the upgrade process by clicking Reset; the device
'burns' the newly loaded files to flash memory and then resets t.he device. After the
device resets, the 'End Process' screen appears displaying the burned configuration
files (refer to the figure below).
Figure 3-106: End Process Wizard Page
10. Click End Process to close the wizard, and then in the 'Enter Network Password'
dialog box, enter your login user name and password (described in ''Accessing the
Web Interface'' on page 20) and click OK; a message box appears informing you of
the new CMP file:
Figure 3-107: Message Box Informing of Upgraded CMP File
11. Click OK; the Web interface now becomes active and reflecting the upgraded device.
3.5.2.4
Backing Up and Restoring Configuration
The 'Configuration File' page allows you to save a copy of the device's current configuration
file modifications as an ini file to a PC. This is useful for backing up your configuration to
protect your device configuration. The saved ini file includes only those parameters that
were modified as well as parameters with other than default values.
In addition, this page allows you to load an ini file to the device. If the device has lost its
configuration, you can restore the device's configuration by loading the previously saved ini
file, or by simply loading a newly created ini file.
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¾ To save and restore the ini file, take these 3 steps:
1.
Open the 'Configuration File' page (Management tab > Software Update menu >
Configuration File).
Figure 3-108: Configuration File Page
2.
3.
3.6
To save the ini file to a PC, perform the following:
a.
Click the Save INI File button; the 'File Download' dialog box opens.
b.
Click the Save button, navigate to the folder in which you want to save the ini file
on your PC, and then click Save; the device copies the ini file to the selected
folder.
To load an ini file to the device, perform the following:
a.
Click the Browse button, navigate to the folder in which the ini file is located,
select the file, and then click Open; the name and path of the file appear in the
field beside the Browse button.
b.
Click the Load INI File button, and then at the prompt, click OK; the device
uploads the ini file and then resets (from the cmp version stored on the flash
memory). Once complete, the 'Enter Network Password' dialog box appears,
requesting you to enter your user name and password.
Status & Diagnostics Tab
The Status & Diagnostics tab on the Navigation bar displays all menus related to the
operating status of the device and device diagnostics. These menus appear in the
Navigation tree and include the following:
„
Status & Diagnostics (refer to ''Status & Diagnostics'' on page 242)
„
Gateway Statistics (refer to ''Gateway Statistics'' on page 248)
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3.6.1
Status & Diagnostics
The Status & Diagnostics menu is used to view and monitor the device's channels, Syslog
messages, hardware and software product information, and to assess the device's statistics
and IP connectivity information. This menu includes the following page items:
3.6.1.1
„
Message Log (refer to ''Viewing the Device's Syslog Messages'' on page 242)
„
Ethernet Port Information (refer to ''Viewing Ethernet Port Information'' on page 243)
„
Active IP Interfaces (refer to ''Viewing Active IP Interfaces'' on page 244)
„
Device Information (refer to ''Viewing Device Information'' on page 244)
„
Performance Statistics (refer to ''Viewing Performance Statistics'' on page 245)
„
Active Alarms (refer to ''Viewing Active Alarms'' on page 245)
„
Trunks & Channels Status (refer to “Viewing Trunks & Channels Status” on page 246)
Viewing the Device's Syslog Messages
The 'Message Log' page displays Syslog debug messages sent by the device. You can
select the Syslog messages in this page, and then copy and paste them into a text editor
such as Notepad. This text file (txt) can then be sent to AudioCodes Technical Support for
diagnosis and troubleshooting.
Note: It's not recommended to keep a Message Log session open for a prolonged
period. This may cause the device to overload. For prolonged (and detailed)
debugging, use an external Syslog server (refer to the Product Reference
Manual).
¾ To activate the Message Log, take these 3 steps:
1.
In the 'Advanced Parameters' page (refer ''Advanced Parameter'' on page 151), set the
parameter 'Debug Level' (or ini file parameter GwDebugLevel) to 6. This parameter
determines the Syslog logging level in the range 0 to 6, where 6 is the highest level.
2.
Open the 'Message Log' page (Status & Diagnostics tab > Status & Diagnostics
menu > Message Log page item); the 'Message Log' page is displayed and the log is
activated.
Figure 3-109: Message Log Screen
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The displayed logged messages are color coded as follows:
3.
•
Yellow - fatal error message
•
Blue - recoverable error message (i.e., non-fatal error)
•
Black - notice message
To clear the page of Syslog messages, in the Navigation tree, click the page item
Message Log again; the page is cleared and new messages begin appearing.
¾ To stop the Message Log, take this step:
„
3.6.1.2
Close the page by accessing any another page in the Web interface.
Viewing the Ethernet Port Information
The 'Ethernet Port Information' page displays read-only information on the Ethernet
connection used by the device. This includes indicating the active port, duplex mode, and
speed. You can also access this page from the 'Home' page (refer to ''Using the Home
Page'' on page 46).
For detailed information on the Ethernet redundancy scheme, refer to ''Ethernet Interface
Redundancy'' on page 380. For detailed information on the Ethernet interface configuration,
refer to ''Ethernet Interface Configuration'' on page 379.
¾ To view Ethernet port information, take the following step:
„
Open the ‘Ethernet Port Information’ page (Status & Diagnostics tab > Status &
Diagnostics menu > Ethernet Port Information page item).
Figure 3-110: Ethernet Port Information Page
Table 3-62: Ethernet Port Information Parameters
Parameter
Description
Active Port
Displays the active Ethernet port (1 or 2).
Port Duplex Mode
Displays the Duplex mode of the Ethernet port (Half Duplex or Full Duplex).
Port Speed
Displays the speed (in Mbps) of the Ethernet port (10 Mbps; 100 Mbps).
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3.6.1.3
Viewing Active IP Interfaces
The 'Active IP Interfaces' page displays the device's IP interfaces, which you configured in
the 'Multiple Interface Table' page (refer to ''Configuring the Multiple Interface Table'' on
page 53) and that are currently active.
¾ To view the 'Active IP Interfaces' page, take this step:
„
Open the 'Active IP Interfaces' page (Status & Diagnostics tab > Status &
Diagnostics menu > Active IP Interfaces page item).
Figure 3-111: Active IP Interfaces Page
3.6.1.4
Viewing Device Information
The 'Device Information' page displays the device's specific hardware and software product
information. This information can help you to expedite troubleshooting. Capture the page
and e-mail it to AudioCodes Technical Support personnel to ensure quick diagnosis and
effective corrective action. This page also displays any loaded files used by the device
(stored in the RAM) and allows you to remove them.
¾ To access the 'Device Information' page, take this step:
„
Open the 'Device Information' page (Status & Diagnostics tab > Status &
Diagnostics menu > Device Information page item).
Figure 3-112: Device Information Page
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The 'Board Type' field number depicts the following devices:
„
Mediant 2000 = 31
„
TP-1610 = 24
¾ To delete any of the loaded files, take this step:
„
3.6.1.5
Click the Delete button corresponding to the files that you want to delete. Deleting a
file takes effect only after the device is reset (refer to ''Resetting the Device'' on page
228).
Viewing Performance Statistics
The 'Performance Statistics' page provides read-only, device performance statistics. This
page is refreshed with new statistics every 60 seconds. The duration that the current
statistics has been collected, is displayed above the statistics table.
¾ To view performance statistics, take the following step:
„
Open the 'Performance Statistics’ page (Status & Diagnostics tab > Status &
Diagnostics menu > Performance Statistics page item).
Figure 3-113: Performance Statistics Page
¾ To reset the performance statistics to zero, take the following step:
„
Version 5.6
Click the Reset Statistics button.
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3.6.1.6
Viewing Active Alarms
The 'Active Alarms' page displays a list of currently active alarms. For each alarm, the
following information is provided:
„
Severity: severity level of the alarm:
•
Critical - alarm displayed in red
•
Major - alarm displayed in orange
•
Minor - alarm displayed in yellow
„
Source: unit from which the alarm was raised
„
Description: brief explanation of the alarm
„
Date: date and time that the alarm was generated
You can also access this page from the 'Home' page (refer to ''Using the Home Page'' on
page 46).
¾ To view the list of alarms, take this step:
„
Open the 'Active Alarms’ page (Status & Diagnostics tab > Status & Diagnostics
menu > Active Alarms page item).
Figure 3-114: Active Alarms Page
3.6.1.7
Viewing Trunks & Channels Status
The 'Trunks & Channels Status' page displays the status of the device's Trunks and the
channels pertaining to these trunks.
¾ To view the status of the device's trunks and the trunks' channels,
take the following step:
„
Open the 'Trunks & Channels Status' page (Status & Diagnostics tab > Status &
Diagnostics menu > Trunks & Channels Status page item).
Figure 3-115: Trunks & Channels Status Page
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Note: The number of trunks and channels displayed on the page depends on the
system configuration.
The page initially displays the first eight trunks and their channels. The page displays eight
consecutive trunks at a time. You can view the next eight trunks, by performing the
procedure below.
¾ To view the next eight trunks, take this step:
„
Click the Go To Page
icon.
Figure 3-116: Example of a Selected Page Icon for Displaying Trunks 17-24
The 'Trunks and Channels Status' page uses the following color-coding icons to indicate the
status of the trunks and channels:
Table 3-63: Color-Coding Icons for Trunk and Channel Status
Trunk
Icon
Color
Description
Gray
Channel
Icon
Color
Description
Disabled
Light Blue
Inactive
Green
Active - OK
Green
Active
Yellow
RAI Alarm
Purple
SS7
Red
LOS/LOF Alarm
Gray
Non Voice
Blue
AIS Alarm
Blue
ISDN Signaling
Orange
D-Channel Alarm
Yellow
CAS Blocked
The 'Trunks & Channels Status' page also allows you to view detailed information regarding
a selected trunk channel, as described in the procedure below.
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¾ To view detailed channel information of a trunk's channel, take
these 2 steps:
1.
Click a required channel pertaining to a trunk for which you want to view information;
the 'Basic Channel Information' page appears, displaying basic information about the
channel:
Figure 3-117: Basic Channel Information Page
2.
3.6.2
To view additional channel information, click the buttons (SIP, Basic, RTP/RTCP, and
Voice Settings) located above on the page.
Gateway Statistics
The 'Gateway Statistics' page allows you to monitor real-time activity such as IP
connectivity information, call details and call statistics, including the number of call
attempts, failed calls, fax calls, etc. This menu includes the following page items:
„
IP to Tel Calls Count and Tel to IP Calls Count (refer to ''Call Counters'' on page 248)
„
Call Routing Status (refer to ''Call Routing Status'' on page 250)
„
SAS/SBC Registered Users (refer to “SAS/SBC Registered Users” on page 251)
„
IP Connectivity (refer to ''IP Connectivity'' on page 252)
Note: The 'Gateway Statistics' pages don't refresh automatically. To view updated
information, re-access the required page.
3.6.2.1
Call Counters
The 'IP to Tel Calls Count' and 'Tel to IP Calls Count' pages provide you with statistical
information on incoming (IP-to-Tel) and outgoing (Tel-to-IP) calls. The statistical information
is updated according to the release reason that is received after a call is terminated (during
the same time as the end-of-call Call Detail Record or CDR message is sent). The release
reason can be viewed in the 'Termination Reason' field in the CDR message.
You can reset the statistical data displayed on the page (i.e., refresh the display), by
clicking the Reset Counters button located on the page.
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¾ To view the IP-to-Tel and Tel-to-IP Call Counters pages, take this
step:
„
Open the Call Counters page that you want to view (Status & Diagnostics tab >
Gateway Statistics menu > IP to Tel Calls Count or Tel to IP Calls Count page
item); the figure below shows the 'IP to Tel Calls Count' page.
Figure 3-118: Calls Count Page
Table 3-64: Call Counters Description
Counter
Description
Number of
Attempted Calls
Indicates the number of attempted calls. It is composed of established and
failed calls. The number of established calls is represented by the 'Number of
Established Calls' counter. The number of failed calls is represented by the
failed-call counters. Only one of the established / failed call counters is
incremented every time.
Number of
Established Calls
Indicates the number of established calls. It is incremented as a result of one of
the following release reasons if the duration of the call is greater than zero:
ƒ
GWAPP_REASON_NOT_RELEVANT (0)
ƒ
GWAPP_NORMAL_CALL_CLEAR (16)
ƒ
GWAPP_NORMAL_UNSPECIFIED (31)
And the internal reasons:
ƒ
RELEASE_BECAUSE_UNKNOWN_REASON
ƒ
RELEASE_BECAUSE_REMOTE_CANCEL_CALL
ƒ
RELEASE_BECAUSE_MANUAL_DISC
ƒ
RELEASE_BECAUSE_SILENCE_DISC
ƒ
RELEASE_BECAUSE_DISCONNECT_CODE
Note: When the duration of the call is zero, the release reason
GWAPP_NORMAL_CALL_CLEAR increments the 'Number of Failed Calls due
to No Answer' counter. The rest of the release reasons increment the 'Number
of Failed Calls due to Other Failures' counter.
Percentage of
Successful Calls
(ASR)
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The percentage of established calls from attempted calls.
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Counter
Description
Number of Calls
Terminated due to
a Busy Line
Indicates the number of calls that failed as a result of a busy line. It is
incremented as a result of the following release reason:
GWAPP_USER_BUSY (17)
Number of Calls
Terminated due to
No Answer
Indicates the number of calls that weren't answered. It's incremented as a
result of one of the following release reasons:
ƒ
GWAPP_NO_USER_RESPONDING (18)
ƒ
GWAPP_NO_ANSWER_FROM_USER_ALERTED (19)
ƒ
GWAPP_NORMAL_CALL_CLEAR (16) (when the call duration is zero)
Number of Calls
Terminated due to
Forward
Indicates the number of calls that were terminated due to a call forward. The
counter is incremented as a result of the following release reason:
RELEASE_BECAUSE_FORWARD
Number of Failed
Calls due to No
Route
Indicates the number of calls whose destinations weren't found. It is
incremented as a result of one of the following release reasons:
ƒ
GWAPP_UNASSIGNED_NUMBER (1)
ƒ
GWAPP_NO_ROUTE_TO_DESTINATION (3)
Number of Failed
Calls due to No
Matched
Capabilities
Indicates the number of calls that failed due to mismatched device capabilities.
It is incremented as a result of an internal identification of capability mismatch.
This mismatch is reflected to CDR via the value of the parameter
DefaultReleaseReason (default is GWAPP_NO_ROUTE_TO_DESTINATION
(3)) or by the GWAPP_SERVICE_NOT_IMPLEMENTED_UNSPECIFIED (79)
reason.
Number of Failed
Calls due to No
Resources
Indicates the number of calls that failed due to unavailable resources or a
device lock. The counter is incremented as a result of one of the following
release reasons:
ƒ
GWAPP_RESOURCE_UNAVAILABLE_UNSPECIFIED
ƒ
RELEASE_BECAUSE_GW_LOCKED
Number of Failed
Calls due to Other
Failures
This counter is incremented as a result of calls that failed due to reasons not
covered by the other counters.
Average Call
Duration (ACD)
[sec]
The average call duration (ACD) in seconds of established calls. The ACD
value is refreshed every 15 minutes and therefore, this value reflects the
average duration of all established calls made within a 15 minute period.
Attempted Fax
Calls Counter
Indicates the number of attempted fax calls.
Successful Fax
Calls Counter
Indicates the number of successful fax calls.
3.6.2.2
Call Routing Status
The 'Call Routing Status' page provides you with information on the current routing method
used by the device. This information includes the IP address and FQDN (if used) of the
Proxy server with which the device currently operates.
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¾ To view the call routing status, take this step:
„
Open the 'Call Routing Status' page (Status & Diagnostics tab > Gateway Statistics
menu > Calls Routing Status page item).
Figure 3-119: Call Routing Status Page
Table 3-65: Call Routing Status Parameters
Parameter
Description
Current Call-Routing
Method
ƒ
Proxy = Proxy server is used to route calls.
ƒ
Routing Table preferred to Proxy = The 'Tel to IP Routing' table (or
'Outbound IP Routing Table' if EnableSBC is set to 1) takes
precedence over a Proxy for routing calls ('Prefer Routing Table'
parameter is set to 'Yes' as described in ''Proxy & Registration
Parameters'' on page 132).
Current Proxy
ƒ
Not Used = Proxy server isn't defined.
ƒ
IP address and FQDN (if exists) of the Proxy server with which the
device currently operates.
ƒ
N/A = Proxy server isn't defined.
ƒ
OK = Communication with the Proxy server is in order.
ƒ
Fail = No response from any of the defined Proxies.
Current Proxy State
3.6.2.3
SAS/SBC Registered Users
The 'SAS Registered Users' page displays a list of up to 250 Stand Alone Survivability
(SAS) and/or IP-to-IP registered users. The SAS feature is configured in the 'SAS
Configuration' page (refer to ''Stand-Alone Survivability'' on page 161). The IP-to-IP feature
is configured (enabled) in the 'SBC Configuration' page (refer to “SBC Configuration” on
page 163).
¾ To view the SAS registered users, take this step:
„
Open the 'SAS Registered Users' page (Status & Diagnostics tab > Gateway
Statistics menu > SAS/SBC Registered Users page item).
Figure 3-120: SAS Registered Users Page
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Table 3-66: SAS Registered Users Parameters
Column Name
Description
Address of
Record
An address-of-record (AOR) is a SIP or SIPS URI that points to a domain with a
location service that can map the URI to another URI (Contact) where the user
might be available.
Contact
SIP URI that can be used to contact that specific instance of the User Agent for
subsequent requests.
3.6.2.4
IP Connectivity
The 'IP Connectivity' page displays online, read-only network diagnostic connectivity
information on all destination IP addresses configured in the 'Tel to IP Routing' page (refer
to ''Tel to IP Routing Table'' on page 175) or 'Outbound IP Routing Table' page if
EnableSBC is set to 1 (refer to “Outbound IP Routing Table” on page 178).
Notes:
•
This information is available only if the parameter 'Enable Alt Routing Tel
to IP' (refer to ''Routing General Parameters'' on page 171) is set to 1
(Enable) or 2 (Status Only).
•
The information in columns 'Quality Status' and 'Quality Info' (per IP
address) is reset if two minutes elapse without a call to that destination.
¾ To view the IP connectivity information, take these 2 steps:
1.
In the 'Routing General Parameters' page, set the parameter 'Enable Alt Routing Tel to
IP' (or ini file parameter AltRoutingTel2IPEnable) to Enable [1] or Status Only [2].
2.
Open the 'IP Connectivity' page (Status & Diagnostics tab > Gateway Statistics
menu > IP Connectivity page item).
Figure 3-121: IP Connectivity Page
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Table 3-67: IP Connectivity Parameters
Column Name
IP Address
Description
The IP address can be one of the following:
ƒ
IP address defined as the destination IP address in the 'Tel to IP Routing'
table (or 'Outbound IP Routing Table' page).
ƒ
IP address resolved from the host name defined as the destination IP
address in the 'Tel to IP Routing' table (or 'Outbound IP Routing Table'
page).
Host Name
Host name (or IP address) as defined in the 'Tel to IP Routing' table (or
'Outbound IP Routing Table' page).
Connectivity
Method
The method according to which the destination IP address is queried
periodically (ICMP ping or SIP OPTIONS request).
Connectivity
Status
The status of the IP address' connectivity according to the method in the
'Connectivity Method' field.
Quality Status
ƒ
OK = Remote side responds to periodic connectivity queries.
ƒ
Lost = Remote side didn't respond for a short period.
ƒ
Fail = Remote side doesn't respond.
ƒ
Init = Connectivity queries not started (e.g., IP address not resolved).
ƒ
Disable = The connectivity option is disabled, i.e., parameter 'Alt Routing Tel
to IP Mode' (AltRoutingTel2IPMode ini) is set to 'None' or 'QoS' (refer to
''Routing General Parameters'' on page 171).
Determines the QoS (according to packet loss and delay) of the IP address.
ƒ
Unknown = Recent quality information isn't available.
ƒ
OK
ƒ
Poor
Notes:
Quality Info.
ƒ
This parameter is applicable only if the parameter 'Alt Routing Tel to IP
Mode' is set to 'QoS' or 'Both' (AltRoutingTel2IPMode = 2 or 3).
ƒ
This parameter is reset if no QoS information is received for 2 minutes.
Displays QoS information: delay and packet loss, calculated according to
previous calls.
Notes:
DNS Status
Version 5.6
ƒ
This parameter is applicable only if the parameter 'Alt Routing Tel to IP
Mode' is set to 'QoS' or 'Both' (AltRoutingTel2IPMode = 2 or 3).
ƒ
This parameter is reset if no QoS information is received for 2 minutes.
DNS status can be one of the following:
ƒ
DNS Disable
ƒ
DNS Resolved
ƒ
DNS Unresolved
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4. ini File Configuration
ini File Configuration
As an alternative to configuring the device using the Web interface (as described in ''WebBased Management'' on page 19), you can configure the device by loading an ini file
containing user-defined parameters. The ini file can be loaded using the following methods:
„
AudioCodes' BootP/TFTP utility (refer to the Product Reference Manual)
„
Any standard TFTP server
„
Web interface (refer to ''Backing Up and Restoring Configuration'' on page 240)
The ini file configuration parameters are saved in the device's non-volatile memory after the
file is loaded to the device. When a parameter is absent from the ini file, the default value is
assigned to that parameter (according to the cmp file loaded to the device) and stored in
the non-volatile memory (thereby, overriding the value previously defined for that
parameter).
Some of the device's parameters are configurable only through the ini file (and not the Web
interface). These parameters usually determine a low-level functionality and are seldom
changed for a specific application.
Notes:
4.1
•
For a list of the ini file parameters, refer to ''The ini File Parameter
Reference'' on page 260. The ini file parameters that are configurable in
the Web interface are described in ''Web-Based Management'' on page
19. The ini parameters that can't be configured using the Web interface
are described in this section.
•
To define or restore default settings using the ini file, refer to ''Default
Settings'' on page 333.
Secured Encoded ini File
The ini file contains sensitive information that is required for the functioning of the device.
Typically, it is loaded to or retrieved from the device using TFTP or HTTP. These protocols
are not secure and vulnerable to potential hackers.
To overcome this security threat, the AudioCodes' TrunkPack Downloadable Conversion
Utility (DConvert) allows you to binary-encode the ini file before loading it to the device
(refer to the Product Reference Manual). If you retrieve an ini file from the device using the
Web interface (refer to Backing Up and Restoring Configuration) that was initially loaded as
encoded to the device, the file is retrieved as encoded and vice versa.
Note: The procedure for loading an encoded ini file is identical to the procedure for
loading an unencoded ini file.
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4.2
The ini File Structure
The ini file can contain any number of parameters. The ini file can contain the following
types of parameters:
4.2.1
„
Individual parameters, which are conveniently grouped (optional) by their functionality
(refer to ''Structure of Individual ini File Parameters'' on page 256)
„
Table parameters, which include multiple individual parameters (refer to ''Structure of
ini File Table Parameters'' on page 257)
Structure Rules
The ini file must adhere to the following format rules:
4.2.2
„
The ini file name must not include hyphens (-) or spaces; if necessary, use an
underscore (_) instead.
„
Lines beginning with a semi-colon (;) are ignored. These can be used for adding
remarks in the ini file.
„
A carriage return (i.e., Enter) must be done at the end of each line.
„
The number of spaces before and after the equals sign (=) is irrelevant.
„
Subsection names for grouping parameters are optional.
„
If there is a syntax error in the parameter name, the value is ignored.
„
Syntax errors in the parameter's value can cause unexpected errors (parameters may
be set to the incorrect values).
„
Parameter string values that denote file names (e.g., CallProgressTonesFileName),
must be enclosed with inverted commas ('…'), e.g., CallProgressTonesFileName =
'cpt_usa.dat'
„
The parameter name is not case-sensitive.
„
The parameter value is not case-sensitive, except for coder names.
„
The ini file must end with at least one carriage return.
Structure of Individual ini File Parameters
The structure of individual ini file parameters in an ini file is shown below:
[Subsection Name]
Parameter_Name = Parameter_Value
Parameter_Name = Parameter_Value
; REMARK
An example of an ini file containing individual ini file parameters is shown below:
[SYSTEM Params]
SyslogServerIP = 10.13.2.69
EnableSyslog = 1
; These are a few of the system-related parameters.
[WEB Params]
LogoWidth = '339'
WebLogoText = 'My Device'
UseWeblogo = 1
; These are a few of the Web-related parameters.
[Files]
CallProgressTonesFileName = 'cpusa.dat'
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4. ini File Configuration
Structure of ini File Table Parameters
You can use anini file to configure table parameters, which include several parameters
(table columns) grouped according to the applications they configure (e.g., NFS and
IPSec). When loading an ini file to the device, it's recommended to include only tables that
belong to applications that are to be configured (dynamic tables of other applications are
empty, but static tables are not).
A table is defined as a secret table (i.e., concealed) if it contains at least one secret data
field or if it depends on another secret table. For example, in the IPSec application, IPSec
tables are defined as secret tables as the IKE table contains a pre-shared key that must be
concealed. Therefore, the SPD table that depends on the IKE table is defined as a secret
table as well. Secret tables are always concealed when loading an ini file to the device.
However, there is a commented title that states that the secret table exists in the device, but
is not to be revealed. Secret tables are always stored in the device's non-volatile memory
and can be overwritten by new tables that are provided in a new ini file. If a secret table
appears in an ini file, it replaces the current table regardless of its content. To delete a
secret table from the device, include an empty table of the same type (with no data lines) as
part of a new ini file.
The ini file table parameter is composed of the following elements:
„
Title of the table: The name of the table in square brackets (e.g.,
[MY_TABLE_NAME]).
„
Format line: Specifies the columns (parameters) of the table (by their string names)
that are to be configured.
„
„
•
The first word of the Format line must be 'FORMAT', followed by the Index field
name, and then an equal (=) sign. After the equal sign, the names of the
parameters (items) are listed.
•
Items must be separated by a comma (,).
•
The Format line must only include columns that can be modified (i.e., parameters
that are not specified as read-only). An exception is Index fields that are always
mandatory.
•
The Format line must end with a semicolon (;).
Data line(s): Contain the actual values of the parameters. The values are interpreted
according to the Format line.
•
The first word of the Data line must be the table’s string name followed by the
Index field.
•
Items must be separated by a comma (,).
•
A Data line must end with a semicolon (;).
End-of-Table Mark: Indicates the end of the table. The same string used for the
table’s title, preceded by a backslash (\), e.g., [\MY_TABLE_NAME].
The following displays an example of the structure of an ini file table parameter.
[Table Title]
; This is the title of the table.
FORMAT Item_Index = Item_Name1, Item_Name2, Item_Name3;
; This is the Format line.
Item 0 = value1, value2, value3;
Item 1 = value1, $$, value3;
; These are the Data lines.
[\Table_Title]
; This is the end-of-the-table-mark.
Refer to the following notes:
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„
Indices (in both the Format and the Data lines) must appear in the same order. The
Index field must never be omitted.
„
The Format line can include a subset of the configurable fields in a table. In this case,
all other fields are assigned with the pre-defined default values for each configured
line.
„
The order of the fields in the Format line isn’t significant (as opposed to the Index
fields). The fields in the Data lines are interpreted according to the order specified in
the Format line.
„
The double dollar sign ($$) in a Data line indicates the default value for the parameter.
„
The order of the Data lines is insignificant.
„
Data lines must match the Format line, i.e., it must contain exactly the same number of
Indices and Data fields and must be in exactly the same order.
„
A line in a table is identified by its table name and Index fields. Each such line may
appear only once in the ini file.
„
Table dependencies: Certain tables may depend on other tables. For example, one
table may include a field that specifies an entry in another table. This method is used
to specify additional attributes of an entity, or to specify that a given entity is part of a
larger entity. The tables must appear in the order of their dependency (i.e., if Table X is
referred to by Table Y, Table X must appear in the ini file before Table Y).
The table below displays an example of an ini file table parameter:
[ PREFIX ]
FORMAT PREFIX_Index = PREFIX_DestinationPrefix,
PREFIX_DestAddress, PREFIX_SourcePrefix, PREFIX_ProfileId,
PREFIX_MeteringCode, PREFIX_DestPort;
PREFIX 0 = 10, 10.13.83.5, *, 0, 255, 0;
PREFIX 1 = 20, 10.13.83.7, *, 0, 255, 0;
PREFIX 2 = 30, 10.13.83.6, *, 0, 255, 0;
PREFIX 3 = 20, 10.13.83.2, *, 0, 255, 0;
[ \PREFIX ]
Note: Do not include read-only parameters in the ini file table parameter, as this can
cause an error when trying to load the file to the device.
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4. ini File Configuration
Example of an ini File
Below is an example of an ini file for the VoIP device.
;Channel Params
DJBufMinDelay = 75
RTPRedundancyDepth = 1
IsProxyUsed = 1
ProxyIP = 192.168.122.179
[CoderName]
FORMAT CoderName_Index = CoderName_Type, CoderName_PacketInterval,
CoderName_rate, CoderName_PayloadType, CoderName_Sce;
CoderName 1= g7231,90
[\CoderName]
;List of serial B-channel numbers
[TrunkGroup]
FORMAT TrunkGroup_Index = TrunkGroup_TrunkGroupNum,
TrunkGroup_FirstTrunkId,TrunkGroup_LastTrunkId,
TrunkGroup_FirstBChannel, TrunkGroup_LastBChannel,
TrunkGroup_FirstPhoneNumber, TrunkGroup_ProfileId,
TrunkGroup_Module;
TrunkGroup 1 = 0,0,0,1,24,1000;
TrunkGroup 2 = 0,1,1,1,24,2000;
TrunkGroup 3 = 0,2,2,1,24,3000;
TrunkGroup 4 = 0,3,3,1,24,4000;
[\TrunkGroup]
CallProgressTonesFilename = 'CPUSA.dat'
SaveConfiguration = 1
4.3
Modifying an ini File
You can modify an ini file currently used by a device. Modifying an ini file instead of loading
an entirely new ini file preserves the device's current configuration, including factory default
values.
¾ To modify an ini file, take these 4 steps:
1.
Save the ini file from the device to your PC using the Web interface (refer to ''Backing
Up and Restoring Configuration'' on page 240).
2.
Open the ini file (using a text file editor such as Microsoft Notepad), and then modify
the ini file parameters according to your requirements.
3.
Save the modified ini file, and then close the file.
4.
Load the modified ini file to the device, using either the BootP/TFTP utility or the Web
interface (refer to ''Backing Up and Restoring Configuration'' on page 240).
Tip:
Version 5.6
Before loading the ini file to the device, verify that the file extension of the ini
file saved on your PC is correct, i.e., *.ini.
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4.4
Reference for ini File Parameters
This subsection lists all the ini file parameters. References to their descriptions in the Web
interface are provided except for those ini file parameters that can only be configured using
the ini file.
4.4.1
Networking Parameters
The networking-related ini file configuration parameters are described in the table below.
Table 4-1: Networking ini File Parameters
Parameter
EthernetPhyConfiguration
Description
Defines the Ethernet connection mode type.
ƒ
[0] = 10Base-T half-duplex
ƒ
[1] = 10Base-T full-duplex
ƒ
[2] = 100Base-TX half-duplex
ƒ
[3] = 100Base-TX full-duplex
ƒ
[4] = Auto-negotiate (default)
For detailed information on Ethernet interface configuration, refer to
''Ethernet Interface Configuration'' on page 379.
DHCPEnable
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
DHCPSpeedFactor
Determines the DHCP renewal speed.
ƒ
[0] = Disable
ƒ
[1] = Normal (default)
ƒ
[2] to [10] = Fast
When set to 0, the DHCP lease renewal is disabled. Otherwise, the
renewal time is divided by this factor. Some DHCP-enabled routers
perform better when set to 4.
EnableDHCPLeaseRenew
al
Enables or disables DHCP renewal support.
ƒ
[0] = Disable (default)
ƒ
[1] = Enable
This parameter is applicable only if DHCPEnable is set to 0 for cases
where booting up the device via DHCP is not desirable, but renewing
DHCP leasing is. When the device is powered up, it attempts to
communicate with a BootP server. If there is no response and if DHCP
is disabled, the device boots from flash. It then attempts to
communicate with the DHCP server to renew the lease.
EnableLANWatchDog
For a description of this parameter, refer to ''General Parameters'' on
page 151.
DNSPriServerIP
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
DNSSecServerIP
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
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Parameter
DNS2IP
Description
This ini file table parameter configures the internal DNS table for
resolving host names into IP addresses. Up to four different IP
addresses (in dotted-decimal notation) can be assigned to a host
name.
The format of this parameter is as follows:
[Dns2Ip]
FORMAT Dns2Ip_Index = Dns2Ip_DomainName,
Dns2Ip_FirstIpAddress, Dns2Ip_SecondIpAddress,
Dns2Ip_ThirdIpAddress, Dns2Ip_FourthIpAddress;
[\Dns2Ip]
For example:
[Dns2Ip]
Dns2Ip 0 = DnsName, 1.1.1.1, 2.2.2.2, 3.3.3.3, 4.4.4.4;
[\Dns2Ip]
Notes:
SRV2IP
ƒ
This parameter can include up to 20 indices.
ƒ
If the internal DNS table is used, the device first attempts to resolve
a domain name using this table. If the domain name isn't found, the
device performs a DNS resolution using an external DNS server.
ƒ
To configure the internal DNS table using the Web interface and for
a description of the parameters in this ini file table parameter, refer
to ''Internal DNS Table'' on page 186.
ƒ
For an explanation on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
This ini file table parameter defines the internal SRV table for resolving
host names to DNS A-Records. Three different A-Records can be
assigned to a host name. Each A-Record contains the host name,
priority, weight, and port. The format of this parameter is as follows:
[SRV2IP]
FORMAT SRV2IP_Index = SRV2IP_InternalDomain,
SRV2IP_TransportType, SRV2IP_Dns1, SRV2IP_Priority1,
SRV2IP_Weight1, SRV2IP_Port1, SRV2IP_Dns2,
SRV2IP_Priority2, SRV2IP_Weight2, SRV2IP_Port2,
SRV2IP_Dns3, SRV2IP_Priority3, SRV2IP_Weight3,
SRV2IP_Port3;
[\SRV2IP]
For example:
[SRV2IP]
SRV2IP 0 =
SrvDomain,0,Dnsname1,1,1,500,Dnsname2,2,2,501,$$,0,0,0;
[\SRV2IP]
Notes:
Version 5.6
ƒ
This parameter can include up to 10 indices.
ƒ
If the Internal SRV table is used, the device first attempts to resolve
a domain name using this table. If the domain name isn't located,
the device performs an SRV resolution using an external DNS
server.
ƒ
To configure the Internal SRV table using the Web interface and for
a description of the parameters in this ini file table parameter, refer
to ''Internal SRV Table'' on page 187.
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Parameter
Description
ƒ
For an explanation on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
EnableSTUN
For a description of this parameter, refer to ''Configuring the
Application Settings'' on page 57.
STUNServerPrimaryIP
For a description of this parameter, refer to ''Configuring the
Application Settings'' on page 57.
STUNServerSecondaryIP
For a description of this parameter, refer to ''Configuring the
Application Settings'' on page 57.
STUNServerDomainName
Defines the domain name for the Simple Traversal of User Datagram
Protocol (STUN) server's address (used for retrieving all STUN
servers with an SRV query). The STUN client can perform the
required SRV query to resolve this domain name to an IP address and
port, sort the server list, and use the servers according to the sorted
list.
Note: Use either the STUNServerPrimaryIP or the
STUNServerDomainName parameter, with priority to the first one.
NATBindingDefaultTimeo
ut
Defines the default NAT binding lifetime in seconds. STUN is used to
refresh the binding information after this time expires.
The valid range is 0 to 2,592,000. The default value is 30.
DisableNAT
Enables / disables the Network Address Translation (NAT)
mechanism.
ƒ
[0] = Enabled.
ƒ
[1] = Disabled (default).
Note: The compare operation that is performed on the IP address is
enabled by default and is controlled by the parameter
EnableIPAddrTranslation. The compare operation that is performed on
the UDP port is disabled by default and is controlled by the parameter
EnableUDPPortTranslation.
EnableIPAddrTranslation
Enables IP address translation.
ƒ
[0] = Disable IP address translation.
ƒ
[1] = Enable IP address translation for RTP, RTCP and T.38
packets (default).
ƒ
[2] = Enable IP address translation for RTP Multiplexing
(ThroughPacket™).
ƒ
[3] = Enable IP address translation for all protocols (RTP, RTCP,
T.38 and RTP Multiplexing).
When enabled, the device compares the source IP address of the first
incoming packet to the remote IP address stated in the opening of the
channel. If the two IP addresses don't match, the NAT mechanism is
activated. Consequently, the remote IP address of the outgoing
stream is replaced by the source IP address of the first incoming
packet.
Notes:
SIP User's Manual
ƒ
The NAT mechanism must be enabled for this parameter to take
effect (DisableNAT set to 0).
ƒ
For information on RTP Multiplexing, refer to ''RTP Multiplexing
(ThroughPacket)'' on page 360.
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Parameter
EnableUDPPortTranslatio
n
Description
ƒ
[0] = Disable UDP port translation (default).
ƒ
[1] = Enable UDP port translation.
When enabled, the device compares the source UDP port of the first
incoming packet, to the remote UDP port stated in the opening of the
channel. If the two UDP ports don't match, the NAT mechanism is
activated. Consequently, the remote UDP port of the outgoing stream
is replaced by the source UDP port of the first incoming packet.
Note: The NAT mechanism and the IP address translation must be
enabled for this parameter to take effect (DisableNAT = 0,
EnableIpAddrTranslation = 1).
NoOpEnable
Enables or disables the transmission of RTP or T.38 No-Op packets.
ƒ
[0] = Disable (default)
ƒ
[1] = Enable
This mechanism ensures that the NAT binding remains open during
RTP or T.38 silence periods.
NoOpInterval
Defines the time interval in which RTP or T.38 No-Op packets are sent
in the case of silence (no RTP / T.38 traffic) when No-Op packet
transmission is enabled.
The valid range is 20 to 65,000 msec. The default is 10,000.
Note: To enable No-Op packet transmission, use the NoOpEnable
parameter.
RTPNoOpPayloadType
Determines the payload type of No-Op packets.
The valid range is 96 to 127 (for the range of Dynamic RTP Payload
Type for all types of non hard-coded RTP Payload types, refer to RFC
3551). The default value is 120.
Note: When defining this parameter, ensure that it doesn't cause
collision with other payload types.
EnableDetectRemoteMAC
Change
Changes the RTP packets according to the MAC address of received
RTP packets and according to Gratuitous Address Resolution Protocol
(GARP) messages.
ƒ
[0] = nothing is changed.
ƒ
[1] = If the device receives RTP packets with a different source
MAC address (than the MAC address of the transmitted RTP
packets), then it sends RTP packets to this MAC address and
removes this IP entry from the device's ARP cache table.
ƒ
[2] = The device uses the received GARP packets to change the
MAC address of the transmitted RTP packets.
ƒ
[3] = both 1 and 2 options above are used (default).
StaticNatIP
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
SyslogServerIP
For a description of this parameter, refer to ''Configuring the
Management Settings'' on page 220.
SyslogServerPort
For a description of this parameter, refer to ''Configuring the
Management Settings'' on page 220.
EnableSyslog
For a description of this parameter, refer to ''Configuring the
Management Settings'' on page 220.
Version 5.6
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Parameter
SyslogOutputMethod
Description
Determines the method used for Syslog messages.
ƒ
[0] = Send all Syslog messages to the defined Syslog server
(default).
ƒ
[1] = Send all Syslog messages using the Debug Recording
mechanism.
ƒ
[2] = Send only Error and Warning level Syslog messages using
the Debug Recording mechanism.
For a detailed description on Debug Recording, refer to Debug
Recording (DR).
BaseUDPport
For a description of this parameter, refer to ''Configuring the RTP /
RTCP Settings'' on page 71.
RemoteBaseUDPPort
For a description of this parameter, refer to “Configuring the RTP /
RTCP Settings” on page 71.
L1L1ComplexTxUDPPort
For a description of this parameter, refer to “Configuring the RTP /
RTCP Settings” on page 71.
L1L1ComplexRxUDPPort
For a description of this parameter, refer to “Configuring the RTP /
RTCP Settings” on page 71.
NTPServerIP
For a description of this parameter, refer to ''Configuring the
Application Settings'' on page 57.
NTPServerUTCOffset
For a description of this parameter, refer to ''Configuring the
Application Settings'' on page 57.
NTPUpdateInterval
For a description of this parameter, refer to ''Configuring the
Application Settings'' on page 57.
IP Routing Table parameters:
The IP routing ini file parameters are array parameters. Each parameter configures a specific column
in the IP routing table. The first entry in each parameter refers to the first row in the IP routing table,
the second entry to the second row and so forth.
In the following example, two rows are configured when the device is in network 10.31.x.x:
RoutingTableDestinationsColumn = 130.33.4.6, 83.4.87.6
RoutingTableDestinationMasksColumn = 255.255.255.255, 255.255.255.0
RoutingTableGatewaysColumn = 10.31.0.1, 10.31.0.112
RoutingTableInterfacesColumn = 0, 1
RoutingTableHopsCountColumn = 20, 20
RoutingTableDestinations
Column
For a description of this parameter, refer to ''Configuring the IP
Routing Table'' on page 62.
RoutingTableDestination
MasksColumn
For a description of this parameter, refer to ''Configuring the IP
Routing Table'' on page 62.
RoutingTableGatewaysCo
lumn
For a description of this parameter, refer to ''Configuring the IP
Routing Table'' on page 62.
RoutingTableHopsCountC
olumn
For a description of this parameter, refer to ''Configuring the IP
Routing Table'' on page 62.
RoutingTableInterfacesCo
lumn
For a description of this parameter, refer to ''Configuring the IP
Routing Table'' on page 62.
VLAN Parameters
VLANMode
SIP User's Manual
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
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Parameter
Description
VLANNativeVLANID
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
VLANOamVLANID
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
VLANControlVLANID
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
VLANMediaVLANID
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
VLANNetworkServiceClas
sPriority
For a description of this parameter, refer to ''Configuring the QoS
Settings'' on page 63.
VLANPremiumServiceCla
ssMediaPriority
For a description of this parameter, refer to ''Configuring the QoS
Settings'' on page 63.
VLANPremiumServiceCla
ssControlPriority
For a description of this parameter, refer to ''Configuring the QoS
Settings'' on page 63.
VlanGoldServiceClassPrio
rity
For a description of this parameter, refer to ''Configuring the QoS
Settings'' on page 63.
VLANBronzeServiceClass
Priority
For a description of this parameter, refer to ''Configuring the QoS
Settings'' on page 63.
EnableDNSasOAM
This parameter applies to both Multiple IPs and VLAN mechanisms.
Multiple IPs: Determines the network type for DNS services.
VLAN: Determines the traffic type for DNS services.
EnableNTPasOAM
VLANSendNonTaggedOn
Native
ƒ
[1] = OAMP (default)
ƒ
[0] = Control.
This parameter applies to both Multiple IPs and VLAN mechanisms.
Multiple IPs: Determines the network type for NTP services.
VLAN: Determines the traffic type for NTP services.
ƒ
[1] = OAMP (default)
ƒ
[0] = Control.
Specify whether to send non-tagged packets on the native VLAN.
ƒ
[0] = Sends priority tag packets (default).
ƒ
[1] = Sends regular packets (with no VLAN tag).
Multiple IPs Parameters
EnableMultipleIPs
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
Note: This parameter is not applicable when configuring multiple
interfaces using the ini file table parameter InterfaceTable.
LocalMediaIPAddress
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
LocalMediaSubnetMask
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
LocalMediaDefaultGW
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
Version 5.6
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Parameter
Description
LocalControlIPAddress
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
LocalControlSubnetMask
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
LocalControlDefaultGW
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
LocalOAMIPAddress
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
LocalOAMSubnetMask
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
LocalOAMDefaultGW
For a description of this parameter, refer to ''Configuring the IP
Settings'' on page 50.
Multiple Interface Table
This ini file table parameter configures the Multiple Interface table for
configuring logical IP addresses. The format of this parameter is as
follows:
[InterfaceTable]
FORMAT InterfaceTable_Index = InterfaceTable_ApplicationTypes,
InterfaceTable_IPv6InterfaceMode, InterfaceTable_IPAddress,
InterfaceTable_PrefixLength, InterfaceTable_Gateway,
InterfaceTable_VlanID, InterfaceTable_InterfaceName;
InterfaceTable 0 = 6, 0, 192.168.85.14, 16, 192.168.0.1, 1, myAll;
[\InterfaceTable]
InterfaceTable
For example:
[InterfaceTable]
FORMAT InterfaceTable_Index = InterfaceTable_ApplicationTypes,
InterfaceTable_IPv6InterfaceMode, InterfaceTable_IPAddress,
InterfaceTable_PrefixLength, InterfaceTable_Gateway,
InterfaceTable_VlanID, InterfaceTable_InterfaceName;
InterfaceTable 0 = 0, 0, 192.168.85.14, 16, 0.0.0.0, 1,
ManagementIF;
InterfaceTable 1 = 2, 0, 200.200.85.14, 24, 0.0.0.0, 200,
myControlIF;
InterfaceTable 2 = 1, 0, 211.211.85.14, 24, 211.211.85.1, 211,
myMediaIF;
[\InterfaceTable]
The above example, configures three network interfaces (OAMP,
Control, and Media applications).
Notes:
SIP User's Manual
ƒ
To configure the Multiple Interface table using the Web interface,
refer to ''Configuring the Multiple Interface Table'' on page 53.
ƒ
For a description of configuring ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
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Parameter
Description
Differential Services.
For detailed information on IP QoS via Differentiated Services, refer to ''IP QoS via Differentiated
Services (DiffServ)'' on page 384.
NetworkServiceClassDiffS
erv
For a description of this parameter, refer to ''Configuring the QoS
Settings'' on page 63.
PremiumServiceClassMed
iaDiffServ
For a description of this parameter, refer to ''Configuring the QoS
Settings'' on page 63.
PremiumServiceClassCon
trolDiffServ
For a description of this parameter, refer to ''Configuring the QoS
Settings'' on page 63.
GoldServiceClassDiffServ
For a description of this parameter, refer to ''Configuring the QoS
Settings'' on page 63.
BronzeServiceClassDiffSe
rv
For a description of this parameter, refer to ''Configuring the QoS
Settings'' on page 63.
NFS Table Parameter
This ini file table parameter defines Network File Systems (NFS) so
that the device can access a remote server's shared files and
directories for loading cmp, ini, and auxiliary files (using the Automatic
Update mechanism). The format of this ini file table parameter is as
follows:
[NFSServers]
FORMAT NFSServers_Index = NFSServers_HostOrIP,
NFSServers_RootPath, NFSServers_NfsVersion,
NFSServers_AuthType, NFSServers_UID, NFSServers_GID,
NFSServers_VlanType;
[\NFSServers]
NFSServers
For example:
[NFSServers]
FORMAT NFSServers_Index = NFSServers_HostOrIP,
NFSServers_RootPath, NFSServers_NfsVersion,
NFSServers_AuthType, NFSServers_UID, NFSServers_GID,
NFSServers_VlanType;
NFSServers 1 = 101.1.13, /audio1, 3, 1, 0, 1, 1;
[\NFSServers]
Notes:
Version 5.6
ƒ
You can configure up to five NFS file systems (0-4).
ƒ
The combination of Host / IP and Root Path must be unique for
each index in the table. For example, the table must include only
one index entry with a Host / IP of '192.168.1.1' and Root Path of
'/audio'.
ƒ
This parameter is applicable only if VLANs are enabled or if
Multiple IPs is configured.
ƒ
To configure NFS using the Web interface and for a description of
the parameters of this ini file table parameter, refer to ''Configuring
the NFS Settings'' on page 60.
ƒ
For a description of configuring ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
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4.4.2
System Parameters
The system-related ini file configuration parameters are described in the table below.
Table 4-2: System ini File Parameters
Parameter
EnableDiagnostics
Description
Checks the correct functionality of the different hardware components
on the device. On completion of the check, if the test fails, the device
sends information on the test results of each hardware component to
the Syslog server.
ƒ
[0] = Rapid and Enhanced self-test mode (default).
ƒ
[1] = Detailed self-test mode (full test of DSPs, PCM, Switch, LAN,
PHY and Flash).
ƒ
[2] = A quicker version of the Detailed self-test mode (full test of
DSPs, PCM, Switch, LAN, PHY, but partial test of Flash).
For detailed information, refer to the Product Reference Manual.
GWAppDelayTime
For a description of this parameter, refer to ''Advanced Parameters''
on page 151.
ActivityListToLog
The Activity Log mechanism enables the device to send log messages
(to a Syslog server) that report certain types of Web actions according
to a pre-defined filter.
The following filters are available:
SIP User's Manual
ƒ
[PVC] (Parameters Value Change) = Changes made on-the-fly to
parameters.
ƒ
[AFL] (Auxiliary Files Loading) = Loading of auxiliary files (e.g., via
'Certificate' screen).
ƒ
[DR] (Device Reset) = Reset of device via the 'Maintenance
Actions' screen.
ƒ
[FB] (Flash Memory Burning) = Burning of files / parameters to
flash (in 'Maintenance Actions' screen).
ƒ
[SWU] (Device Software Update) = cmp loading via the Software
Upgrade Wizard.
ƒ
[ARD] (Access to Restricted Domains) = Access to Restricted
Domains.
The following screens are restricted:
(1) ini parameters (AdminPage)
(2) 'General Security Settings'
(3) 'Configuration File'
(4) 'IPSec/IKE' tables
(5) 'Software Upgrade Key'
(6) 'Internal Firewall'
(7) 'Web Access List'
(8) 'Web User Accounts'
ƒ
[NAA] (Non Authorized Access) = Attempt to access the Web
interface with a false / empty user name or password.
ƒ
[SPC] (Sensitive Parameters Value Change) = Changes made to
sensitive parameters:
(1) IP Address
(2) Subnet Mask
(3) Default Gateway IP Address
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Parameter
Description
(4) ActivityListToLog
For example: ActivityListToLog = 'pvc', 'afl', 'dr', 'fb', 'swu', 'ard', 'naa',
'spc'
ECHybridLoss
Sets the four wire to two wire worst-case Hybrid loss, the ratio
between the signal level sent to the hybrid and the echo level
returning from the hybrid.
ƒ
[0] = 6 dB (default)
ƒ
[1] = N/A
ƒ
[2] = 0 dB
ƒ
[3] = 3 dB
GwDebugLevel
For a description of this parameter, refer to ''Advanced Parameters''
on page 151.
CDRReportLevel
For a description of this parameter, refer to ''Advanced Parameters''
on page 151.
CDRSyslogServerIP
For a description of this parameter, refer to ''Advanced Parameters''
on page 151.
HeartBeatDestIP
Destination IP address (in dotted format notation) to which the device
sends proprietary UDP 'ping' packets.
The default IP address is 0.0.0.0.
HeartBeatDestPort
Destination UDP port to which the heartbeat packets are sent.
The range is 0 to 64000. The default is 0.
HeartBeatIntervalmsec
Delay (in msec) between consecutive heartbeat packets.
EnableRAI
ƒ
[10] = 100000.
ƒ
[-1] = disabled (default).
ƒ
[0] = Disable RAI (Resource Available Indication) service (default).
ƒ
[1] = Enable RAI service.
If RAI is enabled, an SNMP 'acBoardCallResourcesAlarm' Alarm Trap
is sent if device's busy endpoints exceed a predefined (configurable)
threshold.
RAIHighThreshold
High threshold percentage of total calls that are active (busy
endpoints). When the percentage of the device's busy endpoints
exceeds this High Threshold, the device sends the SNMP
acBoardCallResourcesAlarm Alarm Trap with a 'major' Alarm Status.
The range is 0 to 100. The default value is 90.
Note: The percentage of busy endpoints is calculated by dividing the
number of busy endpoints by the total number of “enabled”
endpoints(trunks are physically connected and synchronized with no
alarms and endpoints are defined in the Trunk Group table).
RAILowThreshold
Version 5.6
Low threshold percentage of total calls that are active (busy
endpoints).
When the percentage of the device's busy endpoints falls below this
Low Threshold, the device sends an SNMP
acBoardCallResourcesAlarm Alarm Trap with a 'cleared' Alarm Status.
The range is 0 to 100%. The default value is 90%.
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Parameter
RAILoopTime
Description
Time interval (in seconds) that the device periodically checks call
resource availability.
The valid range is 1 to 200. The default is 10.
Disconnect Supervision Parameters
TelConnectCode
For a description of this parameter, refer to ''Advanced Parameters''
on page 151.
DisconnectOnBrokenCon
nection
For a description of this parameter, refer to ''Advanced Parameters''
on page 151.
BrokenConnectionEventTi
meout
For a description of this parameter, refer to ''Advanced Parameters''
on page 151.
EnableSilenceDisconnect
For a description of this parameter, refer to ''Advanced Parameters''
on page 151.
FarEndDisconnectSilence
Period
For a description of this parameter, refer to ''Advanced Parameters''
on page 151.
FarEndDisconnectSilence
Method
For a description of this parameter, refer to ''Advanced Parameters''
on page 151.
FarEndDisconnectSilence
Threshold
Threshold of the packet count (in percentages) below which is
considered silence by the device.
The valid range is 1 to 100%. The default is 8%.
Note: Applicable only if silence is detected according to packet count
(FarEndDisconnectSilenceMethod = 1).
Automatic Update Parameters
CmpFileURL
Specifies the name of the cmp file and the location of the server (IP
address or FQDN) from which the device loads a new cmp file and
updates itself. The cmp file can be loaded using HTTP, HTTPS, FTP,
FTPS, or NFS.
For example: http://192.168.0.1/filename
Notes:
IniFileURL
ƒ
When this parameter is set in the ini file, the device always loads
the cmp file after it is reset.
ƒ
The cmp file is validated before it's burned to flash. The checksum
of the cmp file is also compared to the previously-burnt checksum
to avoid unnecessary resets.
ƒ
The maximum length of the URL address is 255 characters.
Specifies the name of the ini file and the location of the server (IP
address or FQDN) from which the device loads the ini file. The ini file
can be loaded using: HTTP, HTTPS, FTP, FTPS or NFS.
For example:
http://192.168.0.1/filename
http://192.8.77.13/config<MAC>
https://<username>:<password>@<IP address>/<file name>
Notes:
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ƒ
When using HTTP or HTTPS, the date and time of the ini file are
validated. Only more recently-dated ini files are loaded.
ƒ
The optional string '<MAC>' is replaced with the device's MAC
address. Therefore, the device requests an ini file name that
contains its MAC address. This option enables loading different
configurations for specific devices.
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Parameter
Description
ƒ
PrtFileURL
The maximum length of the URL address is 99 characters.
Specifies the name of the Prerecorded Tones file and the location of
the server (IP address or FQDN) from which it is loaded.
For example: http://server_name/file, https://server_name/file.
Note: The maximum length of the URL address is 99 characters.
CptFileURL
Specifies the name of the CPT file and the location of the server (IP
address or FQDN) from which it is loaded.
For example: http://server_name/file, https://server_name/file.
Note: The maximum length of the URL address is 99 characters.
VpFileURL
Specifies the name of the Voice Prompts file and the location of the
server (IP address or FQDN) from which it is loaded.\
For example: http://server_name/file, https://server_name/file.
Note: The maximum length of the URL address is 99 characters.
CasFileURL
Specifies the name of the CAS file and the location of the server (IP
address or FQDN) from which it is loaded.
For example: http://server_name/file, https://server_name/file.
Note: The maximum length of the URL address is 99 characters.
TLSRootFileUrl
Specifies the name of the TLS trusted root certificate file and the
location URL from where it's downloaded.
TLSCertFileUrl
Specifies the name of the TLS certificate file and the location URL
from where it's downloaded.
UserInfoFileURL
Specifies the name of the User Information file and the location of the
server (IP address or FQDN) from which it is loaded.
For example: http://server_name/file, https://server_name/file
Note: The maximum length of the URL address is 99 characters.
AutoUpdateCmpFile
Enables / disables the Automatic Update mechanism for the cmp file.
ƒ
[0] = The Automatic Update mechanism doesn't apply to the cmp
file (default).
ƒ
[1] = The Automatic Update mechanism includes the cmp file.
AutoUpdateFrequency
Determines the number of minutes the device waits between
automatic updates. The default value is 0 (the update at fixed intervals
mechanism is disabled).
AutoUpdatePredefinedTim
e
Schedules an automatic update to a predefined time of the day.
The range is 'HH:MM' (24-hour format).
For example: 20:18
Note: The actual update time is randomized by five minutes to reduce
the load on the Web servers.
ResetNow
Version 5.6
Invokes an immediate restart of the device. This option can be used to
activate offline (i.e., not on-the-fly) parameters that are loaded via
IniFileUrl.
ƒ
[0] = The immediate restart mechanism is disabled (default).
ƒ
[1] = The device immediately restarts after an ini file with this
parameter set to 1 is loaded.
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Parameter
Description
BootP and TFTP Parameters
The BootP parameters are special 'Hidden' parameters. Once defined and saved in the flash memory,
they are used even if they don't appear in the ini file.
BootPRetries
Note: This parameter only takes effect from the next reset of the
device.
This parameter is used to:
Set the number of BootP requests the
device sends during start-up. The
device stops sending BootP requests
when either BootP reply is received or
number of retries is reached.
BootPSelectiveEnable
Set the number of DHCP
packets the device sends.
After all packets were sent,
if there's still no reply, the
device loads from flash.
ƒ
[1] = 1 BootP retry, 1 sec.
ƒ
[1] = 4 DHCP packets
ƒ
[2] = 2 BootP retries, 3 sec.
ƒ
[2] = 5 DHCP packets
ƒ
[3] = 3 BootP retries, 6 sec.
(default).
ƒ
[3] = 6 DHCP packets
(default)
ƒ
[4] = 10 BootP retries, 30 sec.
ƒ
[4] = 7 DHCP packets
ƒ
[5] = 20 BootP retries, 60 sec.
ƒ
[5] = 8 DHCP packets
ƒ
[6] = 40 BootP retries, 120 sec.
ƒ
[6] = 9 DHCP packets
ƒ
[7] = 100 BootP retries, 300 sec.
ƒ
[7] = 10 DHCP packets
ƒ
[15] = BootP retries indefinitely.
ƒ
[15] = 18 DHCP packets
Enables the Selective BootP mechanism.
ƒ
[1] = Enabled.
ƒ
[0] = Disabled (default).
The Selective BootP mechanism (available from Boot version 1.92)
enables the device's integral BootP client to filter unsolicited
BootP/DHCP replies (accepts only BootP replies that contain the text
'AUDC' in the vendor specific information field). This option is useful in
environments where enterprise BootP/DHCP servers provide
undesired responses to the device's BootP requests.
Note: When working with DHCP (DHCPEnable = 1) the selective
BootP feature must be disabled.
BootPDelay
The interval between the device's startup and the first BootP/DHCP
request that is issued by the device.
ƒ
[1] = 1 second (default).
ƒ
[2] = 3 second.
ƒ
[3] = 6 second.
ƒ
[4] = 30 second.
ƒ
[5] = 60 second.
Note: This parameter only takes effect from the next reset of the
device.
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Parameter
ExtBootPReqEnable
Description
ƒ
[0] = Disable (default).
ƒ
[1] = Enable extended information to be sent in BootP request.
If enabled, the device uses the vendor specific information field in the
BootP request to provide device-related initial startup information such
as blade type, current IP address, software version, etc. For a full list
of the vendor specific Information fields, refer to the Product
Reference Manual.
The BootP/TFTP configuration utility displays this information in the
'Client Info' column (refer to the Product Reference Manual).
Note: This option is not available on DHCP servers.
Serial Parameters
DisableRS232
Enables or disables the device's RS-232 port.
ƒ
[0] = RS-232 serial port is enabled (default).
ƒ
[1] = RS-232 serial port is disabled.
The RS-232 serial port can be used to change the networking
parameters and view error / notification messages. For information on
establishing a serial communications link with the device, refer to the
device's Installation Manual.
SerialBaudRate
Determines the value of the RS-232 baud rate.
The valid range is any value. It is recommended to use the following
standard values: 1200, 2400, 9600 (default), 14400, 19200, 38400,
57600, 115200.
SerialData
Determines the value of the RS-232 data bit.
SerialParity
SerialStop
SerialFlowControl
Version 5.6
ƒ
[7] = 7-bit.
ƒ
[8] = 8-bit (default).
Determines the value of the RS-232 polarity.
ƒ
[0] = None (default).
ƒ
[1] = Odd.
ƒ
[2] = Even.
Determines the value of the RS-232 stop bit.
ƒ
[1] = 1-bit (default).
ƒ
[2] = 2-bit.
Determines the value of the RS-232 flow control.
ƒ
[0] = None (default).
ƒ
[1] = Hardware.
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4.4.3
Web and Telnet Parameters
The Web- and Telnet-related ini file configuration parameters are described in the table
below.
Table 4-3: Web and Telnet ini File Parameters
Parameter
WebAccessList_x
Description
Defines up to ten IP addresses that are permitted to access the
device's Web interface and Telnet interfaces. Access from an
undefined IP address is denied. This security feature is inactive (i.e.,
the device can be accessed from any IP address) when the table is
empty.
For example:
WebAccessList_0 = 10.13.2.66
WebAccessList_1 = 10.13.77.7
The default value is 0.0.0.0 (i.e., the device can be accessed from
any IP address).
For defining the Web and Telnet Access list using the Web interface,
refer to ''Configuring the Web and Telnet Access List'' on page 102.
WebRADIUSLogin
For a description of this parameter, refer to ''Configuring the General
Security Settings'' on page 109.
DisableWebTask
ƒ
[0] = Enable Web management (default).
ƒ
[1] = Disable Web management.
ResetWebPassword
Resets the username and password of the primary and secondary
accounts to their defaults.
ƒ
[0] = Password and username retain their values (default).
ƒ
[1] = Password and username are reset (for the default username
and password, refer to User Accounts).
Note: The username and password cannot be reset from the Web
interface (i.e., via AdminPage or by loading an ini file).
WelcomeMessage
This ini file table parameter configures the Welcome message that
appears after a Web interface login. The format of this parameter is
as follows:
[WelcomeMessage ]
FORMAT WelcomeMessage_Index = WelcomeMessage_Text
WelcomeMessage 1 = "..." ;
WelcomeMessage 2 = "..." ;
WelcomeMessage 3 = "..." ;
[\WelcomeMessage]
For Example:
[WelcomeMessage ]
FORMAT WelcomeMessage_Index = WelcomeMessage_Text
WelcomeMessage 1 = "**********************************" ;
WelcomeMessage 2 = "********* This is a Welcome message ***" ;
WelcomeMessage 3 = "**********************************" ;
[\WelcomeMessage]
Notes:
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ƒ
Each index represents a line of text in the Welcome message box.
Up to 20 indices can be defined.
ƒ
If this parameter is not configured, no Welcome message is
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Parameter
Description
displayed.
ƒ
DisableWebConfig
For a description on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
Determines whether the entire Web interface is in read-only mode.
ƒ
[0] = Enables modifications of parameters (default).
ƒ
[1] = Web interface in read-only mode.
When in read-only mode, parameters can't be modified. In addition,
the following pages can't be accessed: 'Web User Accounts',
'Certificates', 'Regional Settings', 'Maintenance Actions' and all fileloading pages ('Load Auxiliary Files', 'Software Upgrade Wizard', and
'Configuration File').
Note: To return to read/write after you have applied read-only using
this parameter (set to 1), you need to reboot your device with an ini
file that doesn't include this parameter, using the BootP/TFTP Server
utility (refer to the Product Reference Manual).
HTTPport
HTTP port used for Web management (default is 80).
ScenarioFileName
Defines the file name of the Scenario file to be loaded to the device.
The file name must have the dat extension and can be up to 47
characters. For loading a Scenario using the Web interface, refer to
''Loading a Scenario to the Device'' on page 39.
Telnet Parameters
TelnetServerEnable
For a description of this parameter, refer to ''Configuring the
Application Settings'' on page 57.
TelnetServerPort
For a description of this parameter, refer to ''Configuring the
Application Settings'' on page 57.
TelnetServerIdleDisconnect
For a description of this parameter, refer to ''Configuring the
Application Settings'' on page 57.
SSHServerEnable
For a description of this parameter, refer to ''Configuring the
Application Settings'' on page 57.
SSHServerPort
For a description of this parameter, refer to ''Configuring the
Application Settings'' on page 57.
Customizing the Web Appearance Parameters
For detailed information on customizing the Web interface interface, refer to ''Customizing the Web
Interface'' on page 41.
UseProductName
Determines whether the UserProductName text string is displayed
instead of the default product name.
ƒ
[0] = Disabled (default).
ƒ
[1] = Enables the display of the user-defined UserProductName
text string (in the Web interface interface and in the extracted ini
file).
If enabled, the UserProductName text string is displayed instead of
the default product name.
UserProductName
Version 5.6
Text string that replaces the default product name that appears in the
Web interface (upper right-hand corner) and the extracted ini file.
The default is 'Mediant 2000'.
The string can be up to 29 characters.
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Parameter
Description
UseWebLogo
ƒ
[0] = Logo image is used (default).
ƒ
[1] = Text string is used instead of a logo image.
If enabled, AudioCodes' default logo (or any other logo defined by the
LogoFileName parameter) is replaced with a text string defined by
the WebLogoText parameter.
WebLogoText
Text string that replaces the logo image. The string can be up to 15
characters.
LogoWidth
Width (in pixels) of the logo image.
Note: The optimal setting depends on the resolution settings.
The default value is 441, which is the width of AudioCodes' displayed
logo.
Name of the image file (of type GIF, JPEG, or JPG) containing the
user's logo. The logo file name can be used to replace AudioCodes'
default Web logo with a user defined logo.
The file name can be up to 47 characters.
LogoFileName
4.4.4
Security Parameters
The security-related ini file configuration parameters are described in the table below.
Table 4-4: Security ini File Parameters
Parameter
Description
EnableMediaSecurity
For a description of this parameter, refer to ''Configuring Media
Security'' on page 80.
MediaSecurityBehaviour
For a description of this parameter, refer to ''Configuring Media
Security'' on page 80.
SRTPTxPacketMKISize
For a description of this parameter, refer to ''Configuring Media
Security'' on page 80.
RTPAuthenticationDisableTx
For a description of this parameter, refer to ''Configuring Media
Security'' on page 80.
RTPEncryptionDisableTx
For a description of this parameter, refer to ''Configuring Media
Security'' on page 80.
RTCPEncryptionDisableTx
For a description of this parameter, refer to ''Configuring Media
Security'' on page 80.
EnableSIPS
For a description of this parameter, refer to ''General
Parameters'' on page 151.
TLSLocalSIPPort
For a description of this parameter, refer to ''General
Parameters'' on page 151.
TLSVersion
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
TLSReHandshakeInterval
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
SIPSRequireClientCertificate
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
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Parameter
Description
PeerHostNameVerificationMode
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
VerifyServerCertificate
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
TLSRemoteSubjectName
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
OCSPEnable
Enables or disables certificate checking using Online Certificate
Status Protocol (OCSP).
ƒ
[0] = Disable (default).
ƒ
[1] = Enable.
OCSPServerIP
Defines the IP address of the OCSP server.
The default IP address is 0.0.0.0.
OCSPServerPort
Defines the OCSP server's TCP port number.
The default port number is 2560.
OCSPDefaultResponse
Determines the default OCSP behavior when the server cannot
be contacted.
EnableSecureStartup
ƒ
[0] = Rejects peer certificate (default).
ƒ
[1] = Allows peer certificate.
Enables the Secure Startup mode. In this mode, downloading
the *.ini file to the device is restricted to a URL provided in initial
configuration (see parameter IniFileURL) or using DHCP.
ƒ
[0] Disable (default).
ƒ
[1] Enable = disables TFTP and allows secure protocols such
as HTTPS to fetch the device configuration.
Note: For a detailed explanation on Secure Startup, refer to the
Product Reference Manual.
SSHAdminKey
Determines the RSA public key for strong authentication to
logging in to the Secure Shell (SSH) interface (if enabled).
The value should be a base64-encoded string. The value can be
a maximum length of 511 characters.
For additional information, refer to the Product Reference
Manual.
SSHRequirePublicKey
Enables or disables RSA public keys for SSH.
ƒ
[0] = RSA public keys are optional, if a value is configured for
the ini file parameter SSHAdminKey (default).
ƒ
[1] = RSA public keys are mandatory.
IPSec Parameters
EnableIPSec
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
IPSecDPDMode
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
IPSEC_SPD_TABLE
This ini file table parameter configures the IPSec SPD table. The
format of this parameter is as follows:
[IPSEC_SPD_TABLE]
Version 5.6
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Parameter
Description
Format SPD_INDEX = IPSecMode,
IPSecPolicyRemoteIPAddress, IPSecPolicySrcPort,
IPSecPolicyDStPort,IPSecPolicyProtocol,
IPSecPolicyLifeInSec, IPSecPolicyLifeInKB,
IPSecPolicyProposalEncryption_X,
IPSecPolicyProposalAuthentication_X,
IPSecPolicyKeyExchangeMethodIndex,
IPSecPolicyLocalIPAddressType,
IPSecPolicyRemoteTunnelIPAddress,
IPsecPolicyRemoteSubnetMask;
[\IPSEC_SPD_TABLE]
For example:
[IPSEC_SPD_TABLE]
Format SPD_INDEX = IPSecMode,
IPSecPolicyRemoteIPAddress, IpsecPolicySrcPort,
IPSecPolicyDStPort,IPSecPolicyProtocol, IPSecPolicyLifeInSec,
IPSecPolicyProposalEncryption_0,
IPSecPolicyProposalAuthentication_0,
IPSecPolicyProposalEncryption_1,
IPSecPolicyProposalAuthentication_1,
IPSecPolicyKeyExchangeMethodIndex,
IPSecPolicyLocalIPAddressType;
IPSEC_SPD_TABLE 0 = 0, 10.11.2.21, 0, 0, 17, 900, 1,2, 2,2 ,1,
0;
[\IPSEC_SPD_TABLE]
In the example above, all packets designated to IP address
10.11.2.21 that originate from the OAMP interface (regardless of
destination and source ports) and whose protocol is UDP are
encrypted. The IPSec SPD also defines an SA lifetime of 900
seconds and two security proposals (DES/SHA1 and
3DES/SHA1). IPsec is performed using the Transport mode.
Notes:
ƒ
Each row in the table refers to a different IP destination.
ƒ
To support more than one Encryption / Authentication
proposal, for each proposal specify the relevant parameters
in the Format line.
ƒ
The proposal list must be contiguous.
ƒ
To configure the IKE table using the Web interface, refer to
''Configuring the IPSec Table'' on page 114.
ƒ
For an explanation on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
IKE Parameters
IPSec_IKEDB_Table
This ini file table parameter configures the IKE table. The format
of this parameter is as follows:
[IPSec_IKEDB_Table]
Format IKE_DB_INDEX = IKEPolicySharedKey,
IKEPolicyProposalEncryption_X,
IKEPolicyProposalAuthentication_X,
IKEPolicyProposalDHGroup_X, IKEPolicyLifeInSec,
IKEPolicyLifeInKB, IkePolicyAuthenticationMethod;
[\IPSEC_IKEDB_TABLE]
For example:
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Parameter
Description
[IPSec_IKEDB_Table]
Format IKE_DB_INDEX = IKEPolicySharedKey,
IKEPolicyProposalEncryption_0,
IKEPolicypRoposalAuthentication_0,
IKEPolicyProposalDHGroup_0, IKEPolicyProposalEncryption_1,
IKEPolicyProposalAuthentication_1,
IKEPolicyProposalDHGroup_1, IKEPolicyLifeInSec,
IkePolicyAuthenticationMethod;
IPSEC_IKEDB_TABLE 0 = 123456789, 1, 2, 0, 2, 2, 1, 28800, 0;
[\IPSEC_IKEDB_TABLE]
In the example above, a single IKE peer is configured and a preshared key authentication is selected. Its pre-shared key is
123456789. Two security proposals are configured:
DES/SHA1/786DH and 3DES/SHA1/1024DH
Notes:
ƒ
Each row in the table refers to a different IKE peer.
ƒ
To support more than one Encryption / Authentication / DH
Group proposal, for each proposal specify the relevant
parameters in the Format line.
ƒ
The proposal list must be contiguous.
ƒ
To configure the IKE table using the Web interface, refer to
''Configuring the IKE Table'' on page 117.
ƒ
For an explanation on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
Secure Hypertext Transport Protocol (HTTPS) Parameters
HTTPSOnly
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
HTTPSPort
Determines the local Secured HTTPS port of the device.
The valid range is 1 to 65535 (other restrictions may apply within
this range).
The default port is 443.
HTTPSCipherString
Defines the Cipher string for HTTPS (in OpenSSL cipher list
format). For the valid range values, refer to URL
http://www.openssl.org/docs/apps/ciphers.html. The default is
EXP:RC4.
WebAuthMode
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
HTTPSRequireClientCertificate
Requires client certificates for HTTPS connection. The client
certificate must be preloaded to the device, and its matching
private key must be installed on the managing PC. Time and
date must be correctly set on the device, for the client certificate
to be verified.
Version 5.6
ƒ
[0] = Client certificates are not required (default).
ƒ
[1] = Client certificates are required.
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Parameter
HTTPSRootFileName
Description
Defines the name of the HTTPS trusted root certificate file to be
loaded via TFTP. The file must be in base64-encoded PEM
(Privacy Enhanced Mail) format.
The valid range is a 47-character string.
Note: This parameter is only relevant when the device is loaded
via BootP/TFTP. For information on loading this file via the Web
interface, refer to the Product Reference Manual.
HTTPSPkeyFileName
Defines the name of a private key file (in unencrypted PEM
format) to be loaded from the TFTP server.
HTTPSCertFileName
Defines the name of the HTTPS server certificate file to be
loaded via TFTP. The file must be in base64-encoded PEM
format.
The valid range is a 47-character string.
Note: This parameter is only relevant when the device is loaded
using BootP/TFTP. For information on loading this file via the
Web interface, refer to the Product Reference Manual.
Internal Firewall Parameters
This ini file table parameter configures the device's access list
(firewall), which defines network traffic filtering rules. The format
of this parameter is as follows:
[ACCESSLIST]
FORMAT AccessList_Index = AccessList_Source_IP,
AccessList_Net_Mask, AccessList_Start_Port,
AccessList_End_Port, AccessList_Protocol,
AccessList_Packet_Size, AccessList_Byte_Rate,
AccessList_Byte_Burst, AccessList_Allow_Type;
[\ACCESSLIST]
AccessList
For example:
[ACCESSLIST]
FORMAT AccessList_Index = AccessList_Source_IP,
AccessList_Net_Mask, AccessList_Start_Port,
AccessList_End_Port, AccessList_Protocol,
AccessList_Packet_Size, AccessList_Byte_Rate,
AccessList_Byte_Burst, AccessList_Allow_Type;
AccessList 10 = mgmt.customer.com, 255.255.255.255, 0, 80,
tcp, 0, 0, 0, allow;
AccessList 22 = 10.4.0.0, 255.255.0.0, 4000, 9000, any, 0, 0, 0,
block;
[\ACCESSLIST]
In the example above, Rule #10 allows traffic from the host
‘mgmt.customer.com’ destined to TCP ports 0 to 80. Rule #22
blocks traffic from the subnet 10.4.xxx.yyy destined to ports
4000 to 9000.
Notes:
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ƒ
This parameter can include up to 50 indices.
ƒ
If the end of the table is reached without a match, the packet
is accepted.
ƒ
To configure the firewall using the Web interface and for a
description of the parameters of this ini file table parameter,
refer to ''Configuring the Firewall Settings'' on page 103.
ƒ
For a description of configuring with ini file table parameters,
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Parameter
Description
refer to ''Structure of ini File Table Parameters'' on page 257.
For a description of this parameter, refer to ''Configuring the
Firewall Settings'' on page 103.
AccessList_MatchCount
4.4.5
RADIUS Parameters
The RADIUS-related ini file configuration parameters are described in the table below. For
detailed information on the supported RADIUS attributes, refer to ''Supported RADIUS
Attributes'' on page 362.
Table 4-5: RADIUS ini File Parameters
Parameter
Description
EnableRADIUS
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
AAAIndications
For a description of this parameter, refer to ''Configuring RADIUS
Accounting Parameters'' on page 217.
BehaviorUponRadiusTimeout
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
MaxRADIUSSessions
Number of concurrent calls that can communicate with the RADIUS
server (optional).
The valid range is 0 to 240. The default value is 240.
SharedSecret
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
RADIUSRetransmission
Number of retransmission retries.
The valid range is 1 to 10. The default value is 3.
RadiusTO
Determines the time interval (measured in seconds) the device
waits for a response before a RADIUS retransmission is issued.
The valid range is 1 to 30. The default value is 10.
RADIUSAuthServerIP
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
RADIUSAuthPort
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
RADIUSAccServerIP
For a description of this parameter, refer to ''Configuring RADIUS
Accounting Parameters'' on page 217.
RADIUSAccPort
For a description of this parameter, refer to ''Configuring RADIUS
Accounting Parameters'' on page 217.
RadiusAccountingType
For a description of this parameter, refer to ''Configuring RADIUS
Accounting Parameters'' on page 217.
DefaultAccessLevel
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
RadiusLocalCacheMode
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
RadiusLocalCacheTimeout
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
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Parameter
Description
RadiusVSAVendorID
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
RadiusVSAAccessAttribute
For a description of this parameter, refer to ''Configuring the
General Security Settings'' on page 109.
4.4.6
SNMP Parameters
The SNMP-related ini file configuration parameters are described in the table below.
Table 4-6: SNMP ini File Parameters
Parameter
Description
DisableSNMP
For a description of this parameter, refer to ''Configuring the
Management Settings'' on page 220.
SNMPPort
The device's local UDP port used for SNMP Get/Set commands.
The range is 100 to 3999. The default port is 161.
SNMPTrustedMGR_x
Up to five IP addresses of remote trusted SNMP managers from
which the SNMP agent accepts and processes get and set
requests.
Notes:
ƒ
If no values are assigned to these parameters any manager can
access the device.
ƒ
Trusted managers can work with all community strings.
KeepAliveTrapPort
The port to which the keep-alive traps are sent.
The valid range is 0 - 65534. The default is port 162.
SendKeepAliveTrap
When enabled, this parameter invokes the keep-alive trap and
sends it every 9/10 of the time defined in the parameter defining
NAT Binding Default Timeout.
ƒ
[0] = Disable
ƒ
[1] = Enable
SNMPSysOid
Defines the base product system OID.
Default is eSNMP_AC_PRODUCT_BASE_OID_D.
SNMPTrapEnterpriseOid
Defines a Trap Enterprise OID.
Default is eSNMP_AC_ENTERPRISE_OID.
The inner shift of the trap in the AcTrap subtree is added to the end
of the OID in this parameter.
acUserInputAlarmDescriptio
n
Defines the description of the input alarm.
acUserInputAlarmSeverity
Defines the severity of the input alarm.
AlarmHistoryTableMaxSize
Determines the maximum number of rows in the Alarm History
table.
The parameter can be controlled by the Config Global Entry Limit
MIB (located in the Notification Log MIB).
The valid range is 50 to 1000. The default value is 500.
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Parameter
Description
SNMP Trap Parameters
SNMPManagerTableIP_x
For a description of this parameter, refer to ''Configuring the SNMP
Managers Table'' on page 222.
SNMPManagerTrapPort_x
For a description of this parameter, refer to ''Configuring the SNMP
Managers Table'' on page 222.
SNMPManagerTrapUser_x
This parameter can be set to the name of any configured SNMPV3
user to associate with this trap destination. This determines the
trap format, authentication level, and encryption level. By default,
the trap is associated with the SNMP trap community string.
SNMPManagerIsUsed_x
For a description of this parameter, refer to ''Configuring the SNMP
Managers Table'' on page 222.
SNMPManagerTrapSendingE
nable_x
For a description of this parameter, refer to ''Configuring the SNMP
Managers Table'' on page 222.
SNMPTrapManagerHostNam
e
For a description of this parameter, refer to ''Configuring the
Management Settings'' on page 220.
SNMP Community String Parameters
SNMPReadOnlyCommunityS
tring_x
For a description of this parameter, refer to ''Configuring the SNMP
Community Strings'' on page 224.
SNMPReadWriteCommunityS
tring_x
For a description of this parameter, refer to ''Configuring the SNMP
Community Strings'' on page 224.
SNMPTrapCommunityString
For a description of this parameter, refer to ''Configuring the SNMP
Community Strings'' on page 224.
SNMP v3 Users Parameters
SNMPUsers
This ini file table parameter configures SNMP v3 users. The format
of this parameter is as follows:
[SNMPUsers]
FORMAT SNMPUsers_Index = SNMPUsers_Username,
SNMPUsers_AuthProtocol, SNMPUsers_PrivProtocol,
SNMPUsers_AuthKey, SNMPUsers_PrivKey,
SNMPUsers_Group;
[\SNMPUsers]
For example:
[SNMPUsers]
FORMAT SNMPUsers_Index = SNMPUsers_Username,
SNMPUsers_AuthProtocol, SNMPUsers_PrivProtocol,
SNMPUsers_AuthKey, SNMPUsers_PrivKey, SNMPUsers_Group;
SNMPUsers 1 = v3admin1, 1, 0, myauthkey, -, 1;
[\SNMPUsers]
The example above configures user 'v3admin1' with security level
authNoPriv(2), authentication protocol MD5, authentication text
password 'myauthkey', and ReadWriteGroup2.
Notes:
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ƒ
This parameter can include up to 10 indices.
ƒ
To configure SNMP v3 users through the Web interface and for
a description of the parameters of this ini file table, refer to
''Configuring SNMP V3 Users'' on page 225.
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Parameter
Description
For an explanation on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
ƒ
4.4.7
SIP Configuration Parameters
The SIP-related ini file configuration parameters are described in the table below.
Table 4-7: SIP ini File Parameters
Parameter
ReliableConnectionPersistent
Mode
Description
Determines whether all TCP/TLS connections are set as
persistent and therefore, not released.
ƒ
[0] = Disable (default) - all TCP connections (except those that
are set to a proxy IP) are released if not used by any SIP
dialog\transaction.
ƒ
[1] = Enable - TCP connections to all destinations are
persistent and not released unless the device reaches 70% of
its maximum TCP resources.
While trying to send a SIP message connection, reuse policy
determines whether alive connections to the specific destination
are re-used.
Persistent TCP connection ensures less network traffic due to
fewer setting up and tearing down of TCP connections and
reduced latency on subsequent requests due to avoidance of
initial TCP handshake. For TLS, persistent connection may
reduce the number of costly TLS handshakes to establish security
associations, in addition to the initial TCP connection set up.
SIPTransportType
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
TCPLocalSIPPort
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
SIPDestinationPort
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
EnableTCPConnectionReuse
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
SIPTCPTimeout
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
LocalSIPPort
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
EnableFaxReRouting
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
SIPGatewayName
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
IsProxyUsed
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
ProxyName
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
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Parameter
Description
AlwaysSendToProxy
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
PreferRouteTable
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
SIPReroutingMode
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
EnableProxyKeepAlive
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
ProxyKeepAliveTime
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
DNSQueryType
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
ProxyDNSQueryType
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
ProxyIP
This ini file table parameter configures the Proxy Set ID table for
configuring up to six Proxy Sets, each with up to five Proxy server
IP addresses. The format of this parameter is as follows:
[ProxyIP]
FORMAT ProxyIp_Index = ProxyIp_IpAddress,
ProxyIp_TransportType, ProxyIp_ProxySetId;
[\ProxyIP]
For example:
[ProxyIP]
FORMAT ProxyIp_Index = ProxyIp_IpAddress,
ProxyIp_TransportType, ProxyIp_ProxySetId;
ProxyIp 0 = 10.33.37.77, -1, 0;
ProxyIp 1 = 10.8.8.10, 0, 2;
ProxyIp 2 = 10.5.6.7, -1, 1;
[\ProxyIP]
Notes:
ProxySet
ƒ
This parameter can include up to 30 indices (0-29).
ƒ
For assigning various attributes (such as Proxy Load
Balancing) to each Proxy Set ID, refer to the ini file parameter
ProxySet.
ƒ
For configuring the Proxy Set ID table using the Web interface
and for a description of the parameters of this ini file table,
refer to ''Proxy Sets Table'' on page 141.
ƒ
For an explanation on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
This ini file table parameter configures the Proxy Set table by
assigning various attributes per Proxy Set ID. The format of this
parameter is as follows:
[ProxySet]
FORMAT ProxySet_Index = ProxySet_EnableProxyKeepAlive,
ProxySet_ProxyKeepAliveTime,
ProxySet_ProxyLoadBalancingMethod,
ProxySet_IsProxyHotSwap;
[\ProxySet]
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Parameter
Description
For example:
[ProxySet]
FORMAT ProxySet_Index = ProxySet_EnableProxyKeepAlive,
ProxySet_ProxyKeepAliveTime,
ProxySet_ProxyLoadBalancingMethod,
ProxySet_IsProxyHotSwap;
ProxySet 0 = 0, 60, 0, 0;
ProxySet 1 = 1, 60, 1, 0;
[\ProxySet]
Notes:
ƒ
This table parameter can include up to 6 indices (0-5).
ƒ
For configuring the Proxy Sets, refer to the ini file parameter
ProxyIP.
ƒ
For configuring the Proxy Set ID table using the Web interface
and for a description of the parameters of this ini file table,
refer to ''Proxy Sets Table'' on page 141.
ƒ
For an explanation on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
UseSIPTgrp
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
EnableGRUU
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
UserAgentDisplayInfo
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
SIPSDPSessionOwner
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
RetryAfterTime
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
EnablePAssociatedURIHeader
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
EnableContactRestriction
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
RemoveToTagInFailureRespo
nse
Determines whether the device removes the ‘to’ header tag from
final SIP failure responses to INVITE transactions.
ƒ
[0] = Do not remove tag (default).
ƒ
[1] = Remove tag.
ReRegisterOnConnectionFailu
re
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
SourceNumberPreference
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
EnableRTCPAttribute
Enables or disables the use of the 'rtcp' attribute in the outgoing
SDP.
OPTIONSUserPart
SIP User's Manual
ƒ
[0] = Disable
ƒ
[1] = Enable (default)
Defines the User-Part value of the Request-URI for outgoing SIP
OPTIONS requests. If no value is configured, the configuration
parameter ‘Username’ value is used.
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Parameter
Description
A special value is ‘empty’, indicating that no User-Part in the
Request-URI (Host-Part only) is used.
The valid range is a 30-character string. The default value is an
empty string (‘’).
TDMOverIPMinCallsForTrunkAc
tivation
Defines the minimal number of SIP dialogs that must be
established when using TDM Tunneling to consider the specific
trunk as active.
When using TDM Tunneling, if calls from this number of BChannels pertaining to a specific Trunk fail (i.e., SIP dialogs are
not properly set up), an AIS alarm is sent on this trunk toward the
PSTN, and all current calls are dropped. The originator gateway
continues the INVITE attempts. When this number of calls
succeed (i.e., SIP dialogs are set up properly), the AIS alarm is
cleared.
The valid range is 0 to 31. The default value is 0 (i.e., don't send
AIS alarms).
UseGatewayNameForOptions
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
IsProxyHotSwap
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
HotSwapRtx
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
ProxyRedundancyMode
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
ProxyLoadBalancingMethod
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
ProxyIPListRefreshTime
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
IsFallbackUsed
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
UserName
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
Password
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
Cnonce
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
SIPChallengeCachingMode
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
MutualAuthenticationMode
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
IsRegisterNeeded
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
RegistrarIP
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
RegistrarTransportType
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
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Parameter
Description
RegistrarName
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
GWRegistrationName
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
AuthenticationMode
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
OOSOnRegistrationFail
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
RegistrationTime
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
RegistrationTimeDivider
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
RegistrationRetryTime
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
RegisterOnInviteFailure
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
RegistrationTimeThreshold
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
ZeroSDPHandling
Determines the device's response to an incoming SDP with an IP
address of 0.0.0.0 in the Connection line.
ƒ
[0] Sets the IP address of the outgoing SDP Connection line to
0.0.0.0 (default).
ƒ
[1] Sets the IP address of the outgoing SDP Connection line to
the device's own IP address and adds a 'a=sendonly' line to
the SDP.
ForkingHandlingMode
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
Account
This ini file table parameter configures the Account table for
registering and/or authenticating (digest) a Trunk Group (e.g., IPPBX) to a Serving IP Group (e.g., Internet Telephony Service
Provider - ITSP). The format of this parameter is as follows:
[Account]
FORMAT Account_Index = Account_ServedTrunkGroup,
Account_ServedIPGroup, Account_ServingIPGroup,
Account_Username, Account_Password, Account_HostName,
Account_Register, Account_ContactUser;
[\Account]
For example:
[Account]
FORMAT Account_Index = Account_ServedTrunkGroup,
Account_ServedIPGroup, Account_ServingIPGroup,
Account_Username, Account_Password, Account_HostName,
Account_Register, Account_ContactUser;
Account 0 = 1, -1, 1, user, 1234, acl, 1, ITSP1;
[\Account]
Notes:
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ƒ
This table can include up to 10 indices.
ƒ
You can define multiple table indices having the same
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Parameter
Description
ServedTrunkGroup with different ServingIPGroups, username,
password, HostName, and ContactUser. This provides the
capability for registering the same Trunk Group to several
ITSP's (i.e., Serving IP Groups).
IPGroup
ƒ
For configuring the Account table using the Web interface and
for a description of the items in this ini file table, refer to
''Configuring the Account Table'' on page 204.
ƒ
For an explanation on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
This ini file table parameter configures the IP Group table. The
format of this parameter is as follows:
[IPGroup]
FORMAT IPGroup_Index = IPGroup_Type,
IPGroup_Description, IPGroup_ProxySetId,
IPGroup_SIPGroupName, IPGroup_ContactUser,
IPGroup_EnableSurvivability, IPGroup_ServingIPGroup,
IPGroup_SIPReRoutingMode,
IPGroup_AlwaysUseRouteTable;
[\IPGroup]
For example:
[IPGroup]
FORMAT IPGroup_Index = IPGroup_Type, IPGroup_Description,
IPGroup_ProxySetId, IPGroup_SIPGroupName,
IPGroup_ContactUser, IPGroup_EnableSurvivability,
IPGroup_ServingIPGroup, IPGroup_SIPReRoutingMode,
IPGroup_AlwaysUseRouteTable;
IPGroup 1 = 0, "acme gateway", 1, firstIPgroup, , 0, -1, 0, 0;
IPGroup 2 = 0, "abc server", 2, secondIPgroup, , 0, -1, 0, 0;
IPGroup 3 = 0, "IP phones", 1, thirdIPGroup, , 0, -1, 0, 0;
[\IPGroup]
Notes:
ƒ
This table parameter can include up to 9 indices (1-9).
ƒ
For configuring the IP Group table using the Web interface and
for a description of the items in this ini file table, refer to
''Configuring the IP Groups'' on page 201.
ƒ
For an explanation on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
NumberOfActiveDialogs
Defines the maximum number of active SIP dialogs that are not
call related (i.e., REGISTER and SUBSCRIBE). This parameter is
used to control the Registration / Subscription rate.
The valid range is 1 to 20. The default value is 20.
PrackMode
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
AssertedIdMode
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
PAssertedUserName
Defines a 'representative number' (up to 50 characters) that is
used as the User Part of the Request-URI in the P-AssertedIdentity header of an outgoing INVITE (for Tel-to-IP calls).
The default value is NULL.
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Parameter
UseAORInReferToHeader
Description
Defines the source for the SIP URI set in the Refer-To header of
outgoing REFER messages.
ƒ
[0] = Use SIP URI from Contact header of the initial call
(default).
ƒ
[1] = Use SIP URI from To/From header of the initial call.
UseTelURIForAssertedID
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
EnableRPIheader
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
IsUserPhone
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
IsUserPhoneInFrom
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
IsUseToHeaderAsCalledNumb
er
Determines whether the called number is set in the user part of
the To header.
ƒ
[0] = Sets the destination number to the user part of the
Request-URI for IP-to-Tel calls, and sets the Contact header to
the source number for Tel-to-IP calls (default).
ƒ
[1] = Sets the destination number to the user part of the To
header for IP-to-Tel calls, and sets the Contact header to the
username parameter for Tel-to-IP calls.
EnableHistoryInfo
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
SIPSubject
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
MultiPtimeFormat
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
EnableReasonHeader
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
EnableSemiAttendedTransfer
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
SIP183Behavior
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
EnablePtime
Determines whether the ptime header is included in the SDP.
ƒ
[0] = Remove the ptime header from SDP.
ƒ
[1] = Include the ptime header in SDP (default).
EnableUserInfoUsage
For a description of this parameter, refer to ''Advanced
Parameters'' on page 151.
HandleReasonHeader
Determines whether the device uses the value of the incoming
SIP Reason header for Release Reason mapping.
EnableSilenceSuppInSDP
SIP User's Manual
ƒ
[0] Disregard Reason header in incoming SIP messages.
ƒ
[1] Use the Reason header value for Release Reason
mapping (default).
Determines the device's behavior upon receipt of SIP Re-INVITE
messages that include the silencesupp:off attribute.
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Parameter
EnableRport
Description
ƒ
[0] = Disregard the silecesupp attribute (default).
ƒ
[1] = Handle incoming Re-INVITE messages that include the
silencesupp:off attribute in the SDP as a request to switch to
the Voice-Band-Data (VBD) mode.
Enables / disables the usage of the 'rport' parameter in the Via
header.
ƒ
[0] = Enabled.
ƒ
[1] = Disabled (default).
The device adds an 'rport' parameter to the Via header of each
outgoing SIP message. The first Proxy that receives this message
sets the 'rport' value of the response to the actual port from which
the request was received. This method is used, for example, to
enable the device to identify its port mapping outside a NAT.
If the Via doesn't include 'rport' tag, the destination port of the
response is taken from the host part of the Via header.
If the Via includes 'rport' tag without a port value, the destination
port of the response is the source port of the incoming request.
If the Via includes 'rport' tag with a port value (rport=1001), the
destination port of the response is the port indicated in the 'rport'
tag.
DSPVersionTemplateNumber
For a description of this parameter, refer to Configuring the DSP
Templates on page 79.
VBRCoderHeaderFormat
Defines the format of the RTP header for VBR coders.
ƒ
[0] = Payload only (no header, no TOC, no m-factor) -- similar
to RFC 3558 Header Free format (default).
ƒ
[1] = Supports RFC 2658 - 1 byte for interleaving header
(always 0), TOC, no m-factor.
ƒ
[2] = Payload including TOC only, allow m-factor.
ƒ
[3] = RFC 3558 Interleave/Bundled format.
VBRCoderHangover
Determines the required number of silence frames at the
beginning of each silence period, when using the VBR Coder
silence suppression.
The range is 0 to 255. The default value is 1.
AMRFECRedundancyDepth
Defines the AMR / WB-AMR Redundancy depth according to RFC
3267.
The valid range is 0 to 3. The default is 0.
AMRFECNumberOfCodecMod
es
Determines the number of entries to be defined in the AMR
Management Policy table. Each entry defines the policy of a
different rate.
The range is 0 - 9. The default is 0.
AMRFECDelayThreshhold
Defines the one-way delay value (in msec) that may cause the
AMR Hand Out report.
AMRFECDelayHysteresis
Version 5.6
ƒ
0 = 'Hand Out' report is disabled (default)
ƒ
255 msec
Defines the hysteresis of the Delay Threshold for AMR Hand-out
events (in msec). The valid values are 0 to 255. The default is 100
msec.
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Parameter
AMRCoderHeaderFormat
TransparentCoderOnDataCall
Description
Determines the format of the AMR header.
ƒ
[0] (default) = Non standard multiple frames packing in a single
RTP frame. Each frame has a CMR & TOC header.
ƒ
[1] = Reserved.
ƒ
[2] = AMR Header according to RFC 3267 Octet Aligned
header format.
ƒ
[3] = AMR is passed using the AMR IF2 format.
ƒ
[0] = Only use coders from the coder list (default).
ƒ
[1] = Use transparent coder for data calls (according to RFC
4040).
The 'Transparent' coder can be used on data calls. When the
device receives a Setup message from the ISDN with
'TransferCapabilities = data', it can initiate a call using the coder
'Transparent' (even if the coder is not included in the coder list).
The initiated INVITE includes the following SDP attribute:
a=rtpmap:97 CLEARMODE/8000
The default Payload Type is set according to the CoderName
table. If the Transparent coder is not set in the Coders table, the
default value is set to 56. The Payload Type is negotiated with the
remote side, i.e., the selected Payload Type is according to the
remote side selection.
The receiving device must include the 'Transparent' coder in its
coder list.
IsFaxUsed
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
T38UseRTPPort
Defines the port (with relation to RTP port) for sending and
receiving T.38 packets.
ƒ
[0] = Use the RTP port +2 to send / receive T.38 packets
(default).
ƒ
[1] = Use the same port as the RTP port to send / receive T.38
packets.
Notes:
ƒ
For this parameter to take effect, you must reset the device.
ƒ
When the device is configured to use V.152 to negotiate audio
and T.38 coders, the UDP port published in SDP for RTP and
for T38 must be different. Therefore, set the the parameter
T38UseRTPPort to 0.
DefaultReleaseCause
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
IPAlertTimeout
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
SIPPSessionExpires
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
SessionExpiresMethod
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
MINSE
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
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Parameter
Description
SIPMaxRtx
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
SipT1Rtx
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
SipT2Rtx
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
EnableEarlyMedia
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
IgnoreAlertAfterEarlyMedia
Determines the device's interworking of ALERT messages from
PRI to SIP.
ƒ
[0] = Disabled (default).
ƒ
[1] = Enabled.
When enabled, if the device already sent a 183 response with an
SDP included and an ALERT message is received from the Tel
side (with or without Progress Indicator), the device does not send
an additional 18x response and the voice channel remains open.
When disabled, the device sends additional 18x responses as a
result of receiving an ALERT message whether or not a 18x
response was already sent.
EnableTransfer
For a description of this parameter, refer to ''Supplementary
Services'' on page 159.
XferPrefix
For a description of this parameter, refer to ''Supplementary
Services'' on page 159.
EnableMicrosofExt
Modifies the called number for numbers received with Microsoft's
proprietary "ext=xxx" parameter in the SIP INVITE URI user part.
Microsoft Office Communications Server sometimes uses this
proprietary parameter to indicate the extension number of the
called party. For example, if a calling party makes a call to
telephone number 622125519100 Ext. 104, the device receives
the SIP INVITE (from Microsoft's application) with the URI user
part as INVITE sip:622125519100;[email protected] (or INVITE
tel:622125519100;ext=104). If the parameter EnableMicrosofExt
is enabled, the device modifies the called number by adding an
"e" as the prefix, removing the "ext=" parameter, and adding the
extension number as the suffix (e.g., e622125519100104). Once
modified, the device can then manipulate the number further,
using the Number Manipulation tables (refer to ''Number
Manipulation and Routing Parameters'' on page 313) to leave only
the last 3 digits (for example) for sending to a PBX.
XferPrefixIP2Tel
Version 5.6
ƒ
[0] = Disabled (default).
ƒ
[1] = Enabled.
Defines the prefix that is added to the destination number
received in the SIP Refer-to header (in IP-to-Tel calls). This
parameter is applicable for CAS Blind Transfer modes (
TrunkTransferMode = 3).
The valid range is a string of up to 9 characters. The default is an
empty string.
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Parameter
Description
EnableHold
For a description of this parameter, refer to ''Supplementary
Services'' on page 159.
HoldFormat
For a description of this parameter, refer to ''Supplementary
Services'' on page 159.
HeldTimeout
For a description of this parameter, refer to ''Supplementary
Services'' on page 159.
EnableForward
For a description of this parameter, refer to ''Supplementary
Services'' on page 159.
EnableCallWaiting
For a description of this parameter, refer to ''Supplementary
Services'' on page 159.
Send180ForCallWaiting
Determines the SIP response code for indicating call waiting.
ƒ
[0] = Use 182 Queued response to indicate call waiting
(default).
ƒ
[1] = Use 180 Ringing response to indicate call waiting.
HookFlashCode
For a description of this parameter, refer to ''Supplementary
Services'' on page 159.
UseSIPURIForDiversionHeade
r
Sets the URI format in the SIP Diversion header.
ƒ
[0] = 'tel:' (default)
ƒ
[1] = 'sip:'
RTPOnlyModeForTrunk_ID
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
RTPOnlyMode
For a description of this parameter, refer to ''Advanced
Parameters'' on page 151.
TimeoutBetween100And18x
Defines the timeout (in msec) between receiving a 100 Trying
response and a subsequent 18x response. If a 18x response is
not received before this timer expires, the call is disconnected.
The valid range is 0 to 32,000. The default value is 0 (i.e., no
timeout).
TransparentCoderPresentatio
n
Determines the format of the Transparent coder representation in
the SDP.
ƒ
[0] = clearmode (default)
ƒ
[1] = X-CCD
RxDTMFOption
For a description of this parameter, refer to ''DTMF & Dialing
Parameters'' on page 147.
TxDTMFOption
This ini file table parameter determines a single or several (up to
5) preferred transmit DTMF negotiation methods.
The format of this parameter is as follows:
[TxDTMFOption]
FORMAT TxDTMFOption_Index = TxDTMFOption_Type;
[\TxDTMFOption]
For example:
[TxDTMFOption]
TxDTMFOption 0 = 1;
[\TxDTMFOption]
Notes:
ƒ
SIP User's Manual
DTMF negotiation methods are prioritized according to the
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Parameter
Description
order of their appearance.
DisableAutoDTMFMute
ƒ
When out-of-band DTMF transfer is used ([1], [2], or [3]), the
parameter DTMFTransportType is automatically set to 0
(DTMF digits are erased from the RTP stream).
ƒ
When RFC 2833 ([4]) is used, the device:
1) Negotiates RFC 2833 Payload Type (PT) using local and
remote SDPs.
2) Sends DTMF packets using RFC 2833 PT according to the
PT in the received SDP.
3) Expects to receive RFC 2833 packets with the same PT as
configured by the parameter RFC2833PayloadType.
4) Uses the same PT for send and receive if the remote party
doesn't include the RFC 2833 DTMF PT in its SDP.
ƒ
When TxDTMFOption is set to [0], the RFC 2833 PT is set
according to the parameter RFC2833PayloadType for both
transmit and receive.
ƒ
For defining this parameter using the Web interface, refer to
''DTMF & Dialing Parameters'' on page 147.
ƒ
For a description on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
Enables / disables the automatic muting of DTMF digits when outof-band DTMF transmission is used.
ƒ
[0] = Automatic mute is used (default).
ƒ
[1] = No automatic mute of in-band DTMF.
When DisableAutoDTMFMute = 1, the DTMF transport type is set
according to the parameter DTMFTransportType and the DTMF
digits aren't muted if out-of-band DTMF mode is selected
(TxDTMFOption =1, 2 or 3). This enables the sending of DTMF
digits in-band (transparent of RFC 2833) in addition to out-of-band
DTMF messages.
Note: Usually this mode is not recommended.
EnableImmediateTrying
Determines if and when the device sends a 100 Trying response
to an incoming INVITE request.
ƒ
[0] = 100 Trying response is sent upon receipt of Proceeding
message from the PSTN.
ƒ
[1] = 100 Trying response is sent immediately upon receipt of
INVITE request (default).
FirstCallRBTId
For a description of this parameter, refer to ''Advanced
Parameters'' on page 151.
EnableReasonHeader
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
3xxBehavior
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
EnablePChargingVector
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
EnableVMURI
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
Version 5.6
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Parameter
Description
MaxActiveCalls
For a description of this parameter, refer to ''Advanced
Parameters'' on page 151.
MaxCallDuration
For a description of this parameter, refer to ''Advanced
Parameters'' on page 151.
EnableBusyOut
For a description of this parameter, refer to ''Advanced
Parameters'' on page 151.
EnableDigitDelivery2IP
For a description of this parameter, refer to ''Advanced
Parameters'' on page 151.
EnableDigitDelivery
For a description of this parameter, refer to ''Advanced
Parameters'' on page 151.
SITDetectorEnable
Enables or disables Special Information Tone (SIT) detection
according to the ITU-T recommendation E.180/Q.35.
ƒ
[0] = Disable (default).
ƒ
[1] = Enable.
SourceIPAddressInput
For a description of this parameter, refer to ''Routing General
Parameters'' on page 171.
EnableSBC
For a description of this parameter, refer to “SBC Configuration”
on page 163.
SBCRegistrationTime
For a description of this parameter, refer to “SBC Configuration”
on page 163.
Stand-Alone Survivability (SAS) Parameters
EnableSAS
For a description of this parameter, refer to “Stand-Alone
Survivability” on page 161.
SASLocalSIPUDPPort
For a description of this parameter, refer to “Stand-Alone
Survivability” on page 161.
SASDefaultGatewayIP
For a description of this parameter, refer to “Stand-Alone
Survivability” on page 161.
SASRegistrationTime
For a description of this parameter, refer to “Stand-Alone
Survivability” on page 161.
SASLocalSIPTCPPort
For a description of this parameter, refer to “Stand-Alone
Survivability” on page 161.
SASLocalSIPTLSPort
For a description of this parameter, refer to “Stand-Alone
Survivability” on page 161.
SASProxySet
For a description of this parameter, refer to “Stand-Alone
Survivability” on page 161.
RedundantSASProxySet
For a description of this parameter, refer to “Stand-Alone
Survivability” on page 161.
SASSurvivabilityMode
Determines the Survivability mode used by the SAS application.
SIP User's Manual
ƒ
[0] Standard = All incoming INVITE and REGISTER requests
are forwarded to the defined Proxy list in SASProxySet in
Normal mode and handled by the SAS application in
Emergency mode (default).
ƒ
[1] Always Emergency = The SAS application does not use
Keep-Alive messages towards the SASProxySet and instead,
always operates in Emergency mode (as if no Proxy in the
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Parameter
Description
SASProxySet is available).
ƒ
SASBindingMode
SASEnableENUM
SASRegistrationManipulation
[2] Ignore REGISTER = Use regular SAS Normal/Emergency
logic (same as option 0) but when in Normal mode, incoming
REGISTER requests are ignored.
Determines the SAS application database binding mode.
ƒ
[0] URI = If the incoming AoR in the INVITE requests is using a
‘tel:’ URI or ‘user=phone’ is defined, the binding is performed
according to the user part of the URI only. Otherwise, the
binding is according to the entire URI, i.e., User@Host
(default).
ƒ
[1] User Part only = The binding is always performed
according to the User Part only.
Determines whether the SAS application uses ENUM queries to
route incoming INVITE requests when in Emergency mode. Once
an INVITE is received in Emergency mode, the SAS database of
registered users is searched for a matching AoR. If not found, the
Redundant SAS servers are searched. If there is still no match, an
ENUM query is performed and the response is used to correctly
route the INVITE. If no response is received from the ENUM
server, the INVITE is routed to the default gateway.
ƒ
[0] = Disable (default)
ƒ
[1] = Enable
This ini file table parameter is used by the SAS application to
manipulate the User-Part of an incoming REGISTER request AoR
(the To header), before saving it to the registered users database.
The format of this table parameter is as follows:
[SASRegistrationManipulation]
FORMAT SASRegistrationManipulation_Index =
SASRegistrationManipulation_RemoveFromRight,
SASRegistrationManipulation_LeaveFromRight;
[\SASRegistrationManipulation]
ƒ
RemoveFromRight = number of digits removed from the right
side of the User-Part before saving to the registered user
database.
ƒ
LeaveFromRight = number of digits to keep from the right side.
If both RemoveFromRight and LeaveFromRight are defined, the
RemoveFromRight is applied first. The registered database
contains the AoR before and after the manipulation.
The range of both RemoveFromRight and LeaveFromRight is 0 to
30.
Note: This table can include only one index entry.
SASEmergencyNumbers
Version 5.6
Defines emergency numbers for the device's SAS application.
When the device's SAS agent receives a SIP INVITE (from an IP
phone) that includes one of the emergency numbers (in the SIP
user part), it forwards the INVITE to the default gateway
(configured by the parameter SASDefaultGatewayIP), i.e., the
device itself, which sends the call directly to the PSTN. This is
important for routing emergency numbers such as 911 (in North
America) directly to the PSTN. This is applicable to SAS operating
in Normal and Emergency modes.
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Parameter
Description
Up to four emergency numbers can be defined, where each
number can be up to four digits.
Profile Parameters
CoderName
This ini file table parameter defines the device's coder list. This
includes up to five groups of coders (consisting of up to five
coders per group) that can be associated with IP or Tel profiles
('Coder Group Settings' page in the Web interface -- refer to
''Coder Group Settings'' on page 190). The first group of coders
(indices 0 through 4) is the default coder list and default coder
group. The format of this parameter is as follows:
[CoderName]
FORMAT CoderName_Index = CoderName_Type,
CoderName_PacketInterval, CoderName_rate,
CoderName_PayloadType, CoderName_Sce;
[\CoderName]
Where,
ƒ
Type = Coder name
ƒ
PacketInterval = Packetization time
ƒ
Rate = Packetization rate
ƒ
PayloadType = Payload type
ƒ
Sce = Silence suppression mode
For example:
[CoderName]
CoderName 0 = g711Alaw64k, 20,,,0;
CoderName 1 = g726, $$, 3, 38, 0;
CoderName 2 = g729, 40, 255, 255, 1;
[\CoderName]
Notes:
SIP User's Manual
ƒ
This parameter can include up to 25 indices (i.e., five coders
per five coder groups).
ƒ
The coder name is case-sensitive.
ƒ
If silence suppression is not defined for a specific coder, the
value defined by the parameter EnableSilenceCompression is
used.
ƒ
The value of several fields is hard-coded according to common
standards (e.g., payload type of G.711 U-law is always 0).
Other values can be set dynamically. If no value is specified for
a dynamic field, a default value is assigned. If a value is
specified for a hard-coded field, the value is ignored.
ƒ
Only the ptime of the first coder in the defined coder list is
declared in INVITE / 200 OK SDP, even if multiple coders are
defined.
ƒ
If the coder G.729 is selected and silence suppression is
enabled (for this coder), the device includes the string
'annexb=no' in the SDP of the relevant SIP messages. If
silence suppression is set to 'Enable w/o Adaptations',
'annexb=yes' is included. An exception is when the remote
device is a Cisco gateway (IsCiscoSCEMode).
ƒ
Both GSM-FR and MS-GSM coders use Payload Type 3.
When using SDP, it isn’t possible to differentiate between the
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Parameter
Description
two. Therefore, it is recommended not to select both coders
simultaneously.
IPProfile
ƒ
For a list of supported coders, refer to ''Coders'' on page 144.
ƒ
To configure the 'Coders' table in the Web interface, refer to
''Coders'' on page 144.
ƒ
For a description of using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
This ini file table parameter configures the IP profiles table. The
format of this parameter is as follows:
[IPProfile]
FORMAT IPProfile_Index = IPProfile_ProfileName,
IPProfile_IpPreference, IPProfile_CodersGroupID,
IPProfile_IsFaxUsed*, IPProfile_JitterBufMinDelay*,
IPProfile_JitterBufOptFactor*, IPProfile_IPDiffServ*,
IPProfile_SigIPDiffServ*, N/A, IPProfile_RTPRedundancyDepth,
IPProfile_RemoteBaseUDPPort, IPProfile_CNGmode,
IPProfile_VxxTransportType, IPProfile_NSEMode, N/A,
IPProfile_PlayRBTone2IP, IPProfile_EnableEarlyMedia*,
IPProfile_ProgressIndicator2IP*,
IPProfile_EnableEchoCanceller*,
IPProfile_CopyDest2RedirectNumber,
IPProfile_MediaSecurityBehaviour, IPProfile_CallLimit,
IPProfile_ DisconnectOnBrokenConnection;
[\IPProfile]
For example:
[IPProfile]
IPProfile_1 =
name1,2,1,0,10,13,15,44,1,1,6000,0,2,0,0,0,1,0,1,0,0,-1,1;
IPProfile_2 =
name2,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$
$,$$,$$,40,$$;
[\IPProfile]
Notes:
Version 5.6
ƒ
This parameter can appear up to 9 times (i.e., indices 1-9).
ƒ
* Indicates common parameters used in both IP and Tel
profiles.
ƒ
IpPreference = determines the priority of the Profile (1 to 20,
where 20 is the highest preference). If both IP and Tel profiles
apply to the same call, the coders and other common
parameters (indicated with an asterisk) of the preferred Profile
are applied to that call. If the Tel and IP profiles are identical,
the Tel Profile parameters are applied.
ƒ
Two adjacent dollar signs ('$$') indicate that the parameter's
default value is used.
ƒ
IPProfile can be used in the 'Tel to IP Routing' (or 'Outbound IP
Routing Table' if EnableSBC is set to 1) and 'IP to Trunk Group
Routing' tables (Prefix and PSTNPrefix parameters).
ƒ
The 'Profile Name' assigned to a Profile index, must enable
users to identify it intuitively and easily.
ƒ
To configure the IP Profile table using the Web interface, refer
to ''IP Profile Settings'' on page 193.
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Parameter
Description
ƒ
TelProfile
For a description of using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
This ini file table parameter configures the Tel Profile Settings
table. The format of this parameter is as follows:
[TelProfile]
FORMAT TelProfile_Index = TelProfile_ProfileName,
TelProfile_TelPreference, TelProfile_CodersGroupID,
TelProfile_IsFaxUsed*, TelProfile_JitterBufMinDelay*,
TelProfile_JitterBufOptFactor*, TelProfile_IPDiffServ*,
TelProfile_SigIPDiffServ*, TelProfile_DtmfVolume,
TelProfile_InputGain, TelProfile_VoiceVolume, N/A, N/A,
TelProfile_EnableDigitDelivery, TelProfile_EnableEC, N/A, N/A,
TelProfile_FlashHookPeriod, TelProfile_EnableEarlyMedia*,
TelProfile_ProgressIndicator2IP*,
TelProfile_TimeForReorderTone*, N/A, N/A, N/A;
[\TelProfile]
* = Indicates common parameters used in both IP and Tel profiles.
TelPreference = determines the priority of the Profile (1 to 20,
where 20 is the highest preference). If both IP and Tel profiles
apply to the same call, the coders and other common parameters
(indicated with an asterisk) of the preferred Profile are applied to
that call. If the preference of the Tel and IP profiles is identical, the
Tel Profile parameters are applied.
For example:
[TelProfile]
TelProfile 1 =
FaxProfile,1,1,1,40,13,22,33,$$,$$,$$,0,0,0,1,0,0,$$,0,$$,$$,$$,$
$,$$;
TelProfile 2 =
ModemProfile,2,2,0,40,13,$$,$$,$$,$$,$$,$$,$$,$$,0,0,0,$$,0,$$,
$$,$$,$$;
[\TelProfile]
Notes:
SIP User's Manual
ƒ
This parameter can appear up to 9 times (i.e., indices 1-9).
ƒ
Two adjacent dollar signs ('$$') indicates that the parameter's
default value is used.
ƒ
The TelProfile index can be used in the Trunk Group table
(TrunkGroup parameter).
ƒ
The 'Profile Name' assigned to a Profile index must enable
users to identify it intuitively and easily.
ƒ
To configure the Tel Profile table using the Web interface, refer
to ''Tel Profile Settings'' on page 192.
ƒ
For a description of using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
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4. ini File Configuration
Media Server Parameters
The media processing-related ini file configuration parameters are described in the table
below.
Table 4-8: Media Server ini File Parameters
Parameter
AMRCoderHeaderFormat
Description
Determines the format of the AMR header.
ƒ
[0] = Non-standard multiple frames packing in a single RTP frame.
Each frame has a CMR and TOC header.
ƒ
[1] = Reserved.
ƒ
[2] = AMR Header according to RFC 3267 Octet Aligned header
format.
ƒ
[3] = AMR is passed using the AMR IF2 format.
EnableAGC
For a description of this parameter, refer to “Configuring the IPmedia
Settings” on page 76.
AGCGainSlope
For a description of this parameter, refer to “Configuring the IPmedia
Settings” on page 76.
AGCRedirection
For a description of this parameter, refer to “Configuring the IPmedia
Settings” on page 76.
AGCTargetEnergy
For a description of this parameter, refer to “Configuring the IPmedia
Settings” on page 76.
AGCMinGain
Defines the minimum gain (in dB) by the AGC when activated.
The range is 0 to -31. The default is -20.
AGCMaxGain
Defines the maximum gain (in dB) by the AGC when activated.
The range is 0 to 18. The default is 15.
AGCDisableFastAdaptation
Disables the AGC Fast Adaptation mode.
ƒ
[0] = Disable (default)
ƒ
[1] = Enable
AMDDetectionSensitivity
For a description of this parameter, refer to “Configuring the IPmedia
Settings” on page 76.
AMDTimeout
Timeout (in msec) between receiving CONNECT messages from the
ISDN and sending Answering Machine Detection (AMD) results.
The valid range is 1 to 30,000. The default is 2,000 (i.e., 2 seconds).
AMDDetectionDirection
Determines the AMD (Answer Machine Detector) detection direction.
Version 5.6
ƒ
[0] = Detection from the PSTN side
ƒ
[1] = Detection from the IP side
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4.4.9
Voice Mail Parameters
The voice mail-related ini file configuration parameters are described in the table below. For
detailed information on the Voice Mail application, refer to the CPE Configuration Guide for
Voice Mail.
Table 4-9: Voice Mail ini File Parameters
Parameter
Description
VoiceMailInterface
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
SMDI
For a description of this parameter, refer to Configuring the
Voice Mail (VM) Parameters on page 214.
SMDITimeOut
For a description of this parameter, refer to Configuring the
Voice Mail (VM) Parameters on page 214.
LineTransferMode
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
WaitForDialTime
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
MWIOnCode
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
MWIOffCode
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
MWISuffixCode
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
MWISourceNumber
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
Digit Patterns The following digit pattern parameters apply only to VM applications that use the
DTMF communication method. For the available pattern syntaxes, refer to the CPE Configuration
Guide for Voice Mail.
DigitPatternForwardOnBusy
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
DigitPatternForwardOnNoAnswer
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
DigitPatternForwardOnDND
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
DigitPatternForwardNoReason
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
DigitPatternForwardOnBusyExt
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
DigitPatternForwardOnNoAnswerExt
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
DigitPatternForwardOnDNDExt
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
DigitPatternForwardNoReasonExt
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
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Parameter
Description
DigitPatternInternalCall
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
DigitPatternExternalCall
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
TelDisconnectCode
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
DigitPatternDigitToIgnore
For a description of this parameter, refer to ''Configuring
the Voice Mail (VM) Parameters'' on page 214.
4.4.10 PSTN Parameters
The PSTN-related ini file configuration parameters are described in the table below.
Table 4-10: PSTN ini File Parameters
Parameter
Description
PCMLawSelect
For a description of this parameter, refer to “Configuring the TDM Bus
Settings” on page 218.
ProtocolType
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
ProtocolType_x
Same as the description for parameter ProtocolType, but for a specific
trunk ID (x = 0 - 7).
TraceLevel
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
FramingMethod
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
FramingMethod_x
Same as the description for parameter FramingMethod, but for a
specific trunk ID (x = 0 - 7).
TerminationSide
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
TerminationSide_x
Same as the description for parameter TerminationSide, but for a
specific trunk ID (x = 0 - 7).
ClockMaster
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
ClockMaster_x
Same as the description for parameter ClockMaster, but for a specific
trunk ID (x = 0 - 7).
TDMBusClockSource
For a description of this parameter, refer to “Configuring the TDM Bus
Settings” on page 218.
TDMBusPSTNAutoClockE
nable
For a description of this parameter, refer to “Configuring the TDM Bus
Settings” on page 218.
TDMBusLocalReference
For a description of this parameter, refer to “Configuring the TDM Bus
Settings” on page 218.
AutoClockTrunkPriority
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
Version 5.6
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Parameter
Description
TDMBusPSTNAutoClockR
evertingEnable
For a description of this parameter, refer to “Configuring the TDM Bus
Settings” on page 218.
TDMBusEnableFallback
Defines the automatic fallback of the clock.
TDMBusFallbackClock
TDMBusNetrefSpeed
ƒ
[0] = Manual (default)
ƒ
[1] = Auto Non-Revertive
ƒ
[2] = Auto Revertive
Selects the fallback clock source on which the device synchronizes in
the event of a clock failure.
ƒ
[4] = PSTN Network (default)
ƒ
[8] = H.110A
ƒ
[9] = H.110B
ƒ
[10] = NetRef1
ƒ
[11] = NetRef2
Determines the NetRef frequency (for both generation and
synchronization).
ƒ
[0] = 8 kHz (default)
ƒ
[1] = 1.544 MHz
ƒ
[2] = 2.048 MHz
LineCode
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
LineCode_x
Same as the description for parameter LineCode, but for a specific
trunk ID (where 0 depicts the first trunk).
EnableCallingPartyCatego
ry
For a description of this parameter, refer to “Configuring the Digital
Gateway Parameters” on page 207.
BChannelNegotiation
For a description of this parameter, refer to “Configuring the Digital
Gateway Parameters” on page 207.
NFASGroupNumber_x
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
DChConfig_x
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
ISDNNFASInterfaceID_x
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
CASTableIndex_x
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
CASFileName_0
CASFileName_1
CASFileName_7
CAS file name (e.g., 'E_M_WinkTable.dat') that defines the CAS
protocol. It is possible to define up to eight different CAS files by
repeating this parameter. Each CAS file can be associated with one or
more of the device trunks using the parameter CASTableIndex_x.
CASTablesNum
1 to 8. Indicates how many CAS protocol configurations files are
loaded.
IdleABCDPattern
For a description of this parameter, refer to “Configuring the TDM Bus
Settings” on page 218.
IdlePCMPattern
For a description of this parameter, refer to “Configuring the TDM Bus
Settings” on page 218.
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Parameter
Description
LineBuildOut.Loss
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
ISDNRxOverlap_x
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
ISDNRxOverlap
[0] = Disabled (default).
[1] = Enabled.
Any number bigger than one = Number of digits to receive.
Notes:
ƒ
If enabled, the device receives ISDN called number that is sent in
the 'Overlap' mode.
ƒ
The INVITE to IP is sent only after the number (including 'Sending
Complete' Info Element) was fully received (in SETUP and/or
subsequent INFO Q.931 messages).
For detailed information on ISDN overlap dialing, refer to ISDN
Overlap Dialing on page 398.
R2Category
For a description of this parameter, refer to “Configuring the Digital
Gateway Parameters” on page 207.
CallPriorityMode
For a description of this parameter, refer to ''Supplementary Services''
on page 159.
MLPPDefaultNamespace
For a description of this parameter, refer to “Configuring the Digital
Gateway Parameters” on page 207.
SIPDefaultCallPriority
For a description of this parameter, refer to “Configuring the Digital
Gateway Parameters” on page 207.
MLPPDiffserv
For a description of this parameter, refer to ''Supplementary Services''
on page 159.
PreemptionToneDuration
For a description of this parameter, refer to “Configuring the Digital
Gateway Parameters” on page 207.
MLPPNormalizedServiceD
omain
MLPP normalized service domain string. If the device receives an
MLPP ISDN incoming call, it uses the parameter (if different from
‘FFFFFF’) as a Service domain in the SIP Resource-Priority header in
outgoing INVITE messages. If the parameter is ‘FFFFFF’, the
Resource-Priority header is set to the MLPP Service Domain obtained
from the Precedence IE.
The valid value is a 6 hexadecimal digits. The default is ‘000000’.
Note: This parameter is applicable only to device's using the MLPP
NI-2 ISDN variant with CallPriorityMode set to 1.
MLPPDefaultServiceDoma
in
MLPP default service domain string. If the device receives a non
MLPP ISDN incoming call (without a Precedence IE), it uses the
parameter as a Service domain in the SIP Resource-Priority header in
outgoing (Tel-to-IP calls) INVITE messages. This parameter is used in
conjunction with the parameter SipDefaultCallPriority.
The valid value is a 6 hexadecimal digits. The default is "000000".
Note: This parameter is applicable only to device's using the MLPP
NI-2 ISDN variant with CallPriorityMode set to 1.
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Parameter
TrunkAdministrativeState
Description
Defines the administrative state of a trunk.
ƒ
[0] = Lock the trunk; stops trunk traffic to configure the trunk
protocol type.
ƒ
[2] = Unlock the trunk (default); enables trunk traffic.
ISDN Flexible Behavior Parameters
ISDN protocol is implemented in different Switches / PBXs by different vendors. Several
implementations vary a little from the specification. Therefore, to provide a flexible interface that
supports these ISDN variants, the ISDN behavior parameters are used.
ISDNInCallsBehavior
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
ISDNIBehavior
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
ISDNGeneralCCBehavior
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
ISDNOutCallsBehavior
For a description of this parameter, refer to “Configuring the Trunk
Settings” on page 82.
ISDNIBehavior_x
Same as the description for parameter ISDNIBehavior, but for a
specific trunk ID.
ISDNInCallsBehavior_x
Same as the description for parameter ISDNInCallsBehavior, for a
specific trunk ID.
ISDNOutCallsBehavior_x
Same as the description for parameter ISDNOutCallsBehavior, but for
a specific trunk ID.
PlayRBTone2Tel
For a description of this parameter, refer to ''SIP General Parameters''
on page 121.
PlayRBTone2IP
For a description of this parameter, refer to ''SIP General Parameters''
on page 121.
ProgressIndicator2IP
For a description of this parameter, refer to ''Advanced Parameters''
on page 151.
TimeForReorderTone
Busy or Reorder Tone duration that the device, when configured to
protocol type CAS, plays before releasing the line.
The valid range is 0 to 15. The default value is 10 seconds.
Applicable also to ISDN if PlayBusyTone2ISDN = 2. Selection of Busy
or Reorder tone is done according to release cause received from IP.
ISDNDisconnectOnBusyT
one
For a description of this parameter, refer to “Configuring the Digital
Gateway Parameters” on page 207.
DisconnectOnBusyTone
For a description of this parameter, refer to Configuring the Digital
Gateway Parameters on page 207.
EnableVoiceDetection
ƒ
[1] = The device sends 200 OK (to INVITE) messages when
speech/fax/modem is detected from the Tel side.
ƒ
[0] = The device sends 200 OK messages immediately after the
device finishes dialing to the Tel side (default).
Usually this feature is used only when early media (EnableEarlyMedia)
is used to establish voice path before the call is answered.
Notes:
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ƒ
To activate this feature, set EnableDSPIPMDetectors to 1.
ƒ
This feature is applicable only when the protocol type is CAS.
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Parameter
Description
DigitMapping
For a description of this parameter, refer to ''DTMF & Dialing
Parameters'' on page 147.
TimeBetweenDigits
For a description of this parameter, refer to ''DTMF & Dialing
Parameters'' on page 147.
MaxDigits
For a description of this parameter, refer to ''DTMF & Dialing
Parameters'' on page 147.
TimeForDialTone
For a description of this parameter, refer to ''DTMF & Dialing
Parameters'' on page 147.
RegretTime
For a description of this parameter, refer to ''Advanced Parameters''
on page 151.
4.4.11 ISDN and CAS Interworking-Related Parameters
The ISDN- and CAS-related ini file configuration parameters are described in the table
below.
Table 4-11: ISDN and CAS Interworking-Related ini File Parameters
Parameter
Description
EnableTDMoverIP
For a description of this parameter, refer to “Configuring the
Digital Gateway Parameters” on page 207.
EnableISDNTunnelingTel2IP
For a description of this parameter, refer to ''Configuring the
Digital Gateway Parameters'' on page 207.
EnableISDNTunnelingIP2Tel
For a description of this parameter, refer to ''Configuring the
Digital Gateway Parameters'' on page 207.
ISDNDuplicateQ931BuffMode
Controls the activation / deactivation of delivering raw Q.931
messages.
ƒ
[0] = ISDN messages aren't duplicated (default).
ƒ
[128] = All ISDN messages are duplicated.
Note: This parameter is not updated on-the-fly and requires a
device reset.
EnableQSIGTunneling
For a description of this parameter, refer to ''Configuring the
Digital Gateway Parameters'' on page 207.
PlayRBTone2Trunk_ID
For a description of this parameter, refer to ''Configuring the
Trunk Settings'' on page 82.
DigitalOOSBehaviorFor
Trunk_ID
For a description of this parameter, refer to ''Configuring the
Trunk Settings'' on page 82.
DigitalOOSBehavior
For a description of this parameter, refer to ''Configuring the
Digital Gateway Parameters'' on page 207.
RemoveCallingName
For a description of this parameter, refer to ''Configuring the
Digital Gateway Parameters'' on page 207.
DefaultCauseMapISDN2IP
For a description of this parameter, refer to ''Configuring the
Digital Gateway Parameters'' on page 207.
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Parameter
CauseMapSIP2ISDN
Description
This ini file table parameter maps SIP Responses to Q.850
Release Causes. The format of this parameter is as follows:
[CauseMapSIP2ISDN]
FORMAT CauseMapSIP2ISDN_Index =
CauseMapSIP2ISDN_SipResponse,
CauseMapSIP2ISDN_IsdnReleaseCause;
[\CauseMapSIP2ISDN]
Where,
ƒ
SipResponse = SIP Response
ƒ
IsdnReleaseCause = Q.850 Release Cause
For example:
[CauseMapSIP2ISDN]
CauseMapSIP2ISDN 0 = 480,50;
CauseMapSIP2ISDN 0 = 404,3;
[\CauseMapSIP2ISDN]
When a SIP response is received (from the IP side), the device
searches this mapping table for a match. If the SIP response is
found, the Release Cause assigned to it is sent to the PSTN. If
no match is found, the default static mapping is used.
Notes:
CauseMapISDN2SIP
ƒ
This parameter can appear up to 12 times.
ƒ
For an explanation on ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
This ini file table parameter maps Q.850 Release Causes to SIP
Responses.
The format of this parameter is as follows:
[CauseMapISDN2SIP]
FORMAT CauseMapISDN2SIP_Index =
CauseMapISDN2SIP_IsdnReleaseCause,
CauseMapISDN2SIP_SipResponse;
[\CauseMapISDN2SIP]
Where,
ƒ
IsdnReleaseCause = Q.850 Release Cause
ƒ
SipResponse = SIP Response
For example:
[CauseMapISDN2SIP]
CauseMapISDN2SIP 0 = 50,480;
CauseMapISDN2SIP 0 = 6,406;
[\CauseMapISDN2SIP]
When a Release Cause is received (from the PSTN side), the
device searches this mapping table for a match. If the Q.850
Release Cause is found, the SIP response assigned to it is sent
to the IP side. If no match is found, the default static mapping is
used.
Notes:
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ƒ
This parameter can appear up to 12 times.
ƒ
For an explanation on ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
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Parameter
Description
SITQ850Cause
Determines the Q.850 cause value specified in the Reason
header that is included in a 4xx response when Special
Information Tone (SIT) is detected on an IP-to-Tel call.
The valid range is 0 to 127. The default value is 34.
UserToUserHeaderFormat
Determines the format of the User-to-User header.
ƒ
[0] = X-UserToUser (default).
ƒ
[1] = User-to-User with Protocol Discriminator (pd) attribute
User-toUser=3030373435313734313635353b313233343b3834;pd=4
. This is in accordance with the definitions in ‘draft-johnstonsipping-cc-uui-04’.
ƒ
[2] = User-to-User with pd embedded as the first byte.
User-toUser=043030373435313734313635353b313233343b3834;
encoding=hex
RemoveCLIWhenRestricted
For a description of this parameter, refer to ''Configuring the
Digital Gateway Parameters'' on page 207.
ScreeningInd2ISDN
For a description of this parameter, refer to ''Configuring the
Digital Gateway Parameters'' on page 207.
ProgressIndicator2ISDN_ID
For a description of this parameter, refer to ''Configuring the
Trunk Settings'' on page 82.
PIForDisconnectMsg_ID
For a description of this parameter, refer to ''Configuring the
Trunk Settings'' on page 82.
ConnectOnProgressInd
Enables the play of announcements from IP to PSTN without the
need to answer the Tel-to-IP call. It can be used with PSTN
networks that don't support the opening of a TDM channel before
an ISDN Connect message is received.
ƒ
[0] = Connect message isn't sent after SIP 183 Session
Progress message is received (default).
ƒ
[1] = Connect message is sent after SIP 183 Session
Progress message is received.
LocalISDNRBSource_ID
For a description of this parameter, refer to ''Configuring the
Trunk Settings'' on page 82.
PSTNAlertTimeout
For a description of this parameter, refer to ''Advanced
Parameters'' on page 151.
TrunkPSTNAlertTimeout_ID
For a description of this parameter, refer to ''Configuring the
Trunk Settings'' on page 82.
ISDNTransferCapability_ID
For a description of this parameter, refer to ''Configuring the
Trunk Settings'' on page 82.
BChannelNegotiationForTrunk
_ID
For a description of this parameter, refer to ''Configuring the
Trunk Settings'' on page 82.
SendISDNTransferOnConnect
For a description of this parameter, refer to ''Configuring the
Digital Gateway Parameters'' on page 207.
ISDNSubAddressFormat
Determines the format of the 'subaddress' value for ISDN Calling
and Called numbers.
ƒ
Version 5.6
[0] = ASCII (default).
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Parameter
Description
ƒ
[1] = BCD (Binary Coded Decimal)
ƒ
[2] = User Specified
For IP-to-Tel calls, if the incoming SIP INVITE message includes
subaddress values in the 'isub' parameter for the Called Number
(in the Request-URI) and/or the Calling Number (in the From
header), these values are mapped to the outgoing ISDN SETUP
message.
If the incoming ISDN SETUP message includes 'subaddress'
values for the Called Number and/or the Calling Number, these
values are mapped to the outgoing SIP INVITE message's ‘isub’
parameter in accordance with RFC 4715.
EnableHold2ISDN
For a description of this parameter, refer to ''Supplementary
Services'' on page 159.
EnableUUITel2IP
For a description of this parameter, refer to ''Configuring the
Digital Gateway Parameters'' on page 207.
EnableUUIIP2Tel
For a description of this parameter, refer to ''Configuring the
Digital Gateway Parameters'' on page 207.
ScreeningInd2IP
For a description of this parameter, refer to ''Configuring the
Digital Gateway Parameters'' on page 207.
SupportRedirectInFacility
Determines whether the Redirect Number is retrieved from the
Facility IE.
ƒ
[0] = Not supported (default).
ƒ
[1] = Supports partial retrieval of Redirect Number (number
only) from the Facility IE in ISDN SETUP messages.
Applicable to Redirect Number according to ECMA-173 Call
Diversion Supplementary Services.
Note: To enable this feature, ISDNDuplicateQ931BuffMode must
be set to 1.
EnableCIC
Determines whether Carrier Identification Code (CIC) is relayed
to ISDN.
ƒ
[0] = Do not relay the Carrier Identification Code (CIC) to
ISDN (default).
ƒ
[1] = CIC is relayed to the ISDN in Transit Network Selection
(TNS) IE.
If enabled, the CIC code (received in an INVITE Request-URI) is
included in a TNS IE in the ISDN SETUP message.
For example: INVITE sip:555666;[email protected] sip/2.0.
Note: Currently, this feature is supported only in the SIP-to-ISDN
direction.
EnableAOC
ƒ
[0] = Not used (default).
ƒ
[1] = ISDN Advice of Charge (AOC) messages are
interworked to SIP.
The device supports receipt of ISDN (Euro ISDN) AOC
messages. AOC messages can be received during a call
(FACILITY messages) or at the end of a call (DISCONNECT or
RELEASE messages). The device converts the AOC messages
into SIP INFO (during a call) and BYE (end of a call) messages,
using a proprietary AOC SIP header. The device supports both
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Parameter
Description
Currency and Pulse AOC messages.
PlayBusyTone2ISDN
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
TrunkTransferMode_X
Determines the supported trunk transfer method when a SIP
REFER message is received.
ƒ
[0] = Not supported (default).
ƒ
[1] = Supports CAS NFA DMS-100 transfer. When a SIP
REFER message is received, the device performs a Blind
Transfer by executing a CAS Wink, waits for an acknowledged
Wink from the remote side, dials the Refer-to number to the
switch, and then releases the call.
Note: A specific NFA CAS table is required.
ƒ
[2] = Supports ISDN transfer: RLT (DMS-100), TBCT (NI2),
ECT (EURO ISDN), and Path Replacement (QSIG). When a
SIP REFER message is received, the device performs a
transfer by sending FACILITY messages to the PBX with the
necessary information on the call's legs that are to be
connected. The different ISDN variants use slightly different
methods (using FACILITY messages) to perform the transfer.
ƒ
[3] = Supports CAS Normal transfer. When a SIP REFER
message is received, the device performs a Blind Transfer by
executing a CAS Wink, dialing the Refer-to number to the
switch, and then releasing the call.
ƒ
[4] = Supports QSIG Single Step transfer:
IP-to-Tel: When a SIP REFER message is received, the
device performs a transfer by sending a FACILITY message to
the PBX, initiating Single Step transfer. Once a success return
result is received, the transfer is completed.
Tel-to-IP: When a FACILITY message initiating Single Step
transfer is received from the PBX, a REFER message is sent
to the IP side.
Notes:
ƒ
To use QSIG Path Replacement, the parameter
UserToUserHeaderFormat must be set to 1.
ƒ
To configure Trunk Transfer Mode using the Web interface,
refer to ''Configuring the Trunk Settings'' on page 82.
CASTransportType
For a description of this parameter, refer to ''Configuring the
Voice Settings'' on page 66.
CASAddressingDelimiters
Determines if delimiters are added to the dialed address or dialed
ANI parameters.
ƒ
[0] = Disable (default)
ƒ
[1] = Enable
When this parameter is enabled, delimiters such as '*', '#', and
'ST' are added to the dialed address or dialed ANI parameters.
When it is disabled, the address and ANI strings remain without
delimiters.
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Parameter
CASDelimitersPaddingUsage
Description
Defines the digits string delimiter padding usage per trunk.
ƒ
[0] (default) = default address string padding: '*XXX#' (where
XXX is the digit string that begins with '*' and ends with '#',
when using padding).
ƒ
[1] = special use of asterisks delimiters: '*XXX*YYY*' (where
XXX is the address, YYY is the source phone number, and '*'
is the only delimiter padding).
CasStateMachineGenerateDigi
tOnTime
For a description of this parameter, refer to ''Configuring the CAS
State Machines'' on page 97.
CasStateMachineGenerateInte
rDigitTime
For a description of this parameter, refer to ''Configuring the CAS
State Machines'' on page 97.
CasStateMachineDTMFMaxOn
DetectionTime
For a description of this parameter, refer to ''Configuring the CAS
State Machines'' on page 97.
CasStateMachineDTMFMinOn
DetectionTime
For a description of this parameter, refer to ''Configuring the CAS
State Machines'' on page 97.
CasStateMachineMaxNumOfIn
comingAddressDigits
For a description of this parameter, refer to ''Configuring the CAS
State Machines'' on page 97.
CasStateMachineMaxNumOfIn
comingANIDigits
For a description of this parameter, refer to ''Configuring the CAS
State Machines'' on page 97.
CasStateMachineCollectANI
For a description of this parameter, refer to ''Configuring the CAS
State Machines'' on page 97.
CasStateMachineDigitSignalin
gSystem
For a description of this parameter, refer to ''Configuring the CAS
State Machines'' on page 97.
EnableDSPIPMDetectors
Enables or disables the device's DSP detectors.
ƒ
[0] = Disable (default).
ƒ
[1] = Enable.
Notes:
ƒ
The device's Feature Key should contain the 'IPMDetector'
DSP option.
ƒ
When enabled (1), the number of available channels is
reduced by a factor of 5/6. For example, a device with 8 E1
spans, capacity is reduced to 6 spans (180 channels), while a
device with 8 T1 spans, capacity remains the same (192
channels).
XChannelHeader
For a description of this parameter, refer to ''Advanced
Parameters'' on page 151.
AddIEinSetup
For a description of this parameter, refer to ''Configuring the
Digital Gateway Parameters'' on page 207.
SendIEonTG
For a description of this parameter, refer to ''Configuring the
Digital Gateway Parameters'' on page 207.
ISDNDMSTimerT310
Overrides the T310 timer for the DMS-100 ISDN variant.
T310 defines the timeout between the reception of a
PROCEEDING message and the reception of an ALERTING /
CONNECT message.
The valid range is 10 to 30. The default value is 10 (seconds).
Note: Applicable only to Nortel DMS and Nortel MERIDIAN PRI
variants (ProtocolType = 14 and 35).
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Parameter
ISDNJapanNTTTimerT3JA
Description
T3_JA timer (in seconds). This parameter overrides the internal
PSTN T301 timeout on the Users Side (TE side).
If an outgoing call from the device to ISDN is not answered during
this timeout, the call is released.
The valid range is 10 to 240. The default value is 50.
Applicable only to Japan NTT PRI variant (ProtocolType = 16).
Note: This timer is also affected by the parameter
PSTNAlertTimeout.
EnablePatternDetector
For a description of this parameter, refer to “Configuring the
Digital Gateway Parameters” on page 207.
PDPattern
Defines the patterns that can be detected by the Pattern
Detector.
The valid range is 0 to 0xFF.
PDThreshold
Defines the number of consecutive patterns to trigger the pattern
detection event.
The valid range is 0 to 31. The default is 5.
Enable911LocationIdIP2Tel
Enables interworking of Emergency Location Identification from
SIP to PRI.
ƒ
[0] = Disabled (default)
ƒ
[1] = Enabled
When enabled, the From header received in the SIP INVITE is
translated into the following ISDN Information Elements (IE):
ƒ
Emergency Call Control IE.
ƒ
Generic Information IE to carry the Location Identification
Number information.
ƒ
Generic Information IE to carry the Calling Geodetic Location
information.
Note: This capability is supported only for the NI-2 ISDN variant.
EarlyAnswerTimeout
Version 5.6
Defines the time (in seconds) that the device waits for a
CONNECT message from the called party (Tel side) after sending
a SETUP message. If the timer expires, the call is answered by
sending a 200 OK message (IP side).
The valid range is 0 to 600. The default value is 0 (i.e., disabled).
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4.4.12 Number Manipulation and Routing Parameters
The number manipulation and routing-related ini file configuration parameters are described
in the table below.
Table 4-12: Number Manipulation and Routing ini File Parameters
Parameter
TrunkGroup
Description
This ini file table parameter defines the device's Trunks and
assigns them to Trunk Groups. The format of this parameter is
shown below:
[TrunkGroup]
FORMAT TrunkGroup_Index = TrunkGroup_TrunkGroupNum,
TrunkGroup_FirstTrunkId, TrunkGroup_LastTrunkId,
TrunkGroup_FirstBChannel, TrunkGroup_LastBChannel,
TrunkGroup_FirstPhoneNumber, TrunkGroup_ProfileId,
TrunkGroup_Module;
[\TrunkGroup]
For example:
[TrunkGroup]
TrunkGroup 1 = 0, 0, 0, 1, 31 ,401, 0; (E1 span)
TrunkGroup 1 = 0, 0, 0, 1, 31, $$, 1;
TrunkGroup 2 = 1, 2, 2, 1, 24, 3000;
(T1 span)
TrunkGroup 1 = 2, 0, 7, 1, 20, 1000;
(8 E1 spans; 20 Bchannels)
TrunkGroup 1 = 0, 0, 3, *, 1000;(4 E1 spans; all B-channels)
[\TrunkGroup]
Notes:
ƒ
The parameter TrunkGroup_Module is not applicable.
ƒ
To represent all B-channels, use an asterisk (*).
ƒ
For configuring this table in the Web interface, refer to
Configuring the Trunk Group Table on page 195.
ƒ
For a description of ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
DefaultNumber
For a description of this parameter, refer to ''DTMF & Dialing
Parameters'' on page 147.
ChannelSelectMode
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
TrunkGroupSettings
This ini file table parameter defines rules for port allocation per
Trunk Group. If no rule exists, the global rule defined by the
parameter ChannelSelectMode takes effect. The format of this
parameter is as follows:
[TrunkGroupSettings]
FORMAT TrunkGroupSettings_Index =
TrunkGroupSettings_TrunkGroupId,
TrunkGroupSettings_ChannelSelectMode,
TrunkGroupSettings_RegistrationMode,
TrunkGroupSettings_GatewayName,TrunkGroupSettings_Cont
actUser, TrunkGroupSettings_ServingIPGroup;
[\TrunkGroupSettings]
For example:
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Parameter
Description
[TrunkGroupSettings]
TrunkGroupSettings 0 = 1, 0, 5, audiocodes, user, 1;
TrunkGroupSettings 1 = 2, 1, 0, localname, user1, 2;
[\TrunkGroupSettings]
Notes:
ƒ
This parameter can include up to 240 indices.
ƒ
For configuring Trunk Group Settings using the Web
interface, refer to ''Configuring Trunk Group Settings'' on page
197.
ƒ
For a description on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
AddTrunkGroupAsPrefix
For a description of this parameter, refer to ''Routing General
Parameters'' on page 171.
AddPortAsPrefix
For a description of this parameter, refer to ''Routing General
Parameters'' on page 171.
ReplaceEmptyDstWithPortNum
ber
For a description of this parameter, refer to Routing General
Parameters on page 171.
CopyDestOnEmptySource
ƒ
[0] = Leave Source Number empty (default).
ƒ
[1] = If the Source Number of a Tel-to-IP call is empty, the
Destination Number is copied to the Source Number.
AddNPIandTON2CallingNumbe
r
For a description of this parameter, refer to Routing General
Parameters on page 171.
AddNPIandTON2CalledNumber
For a description of this parameter, refer to Routing General
Parameters on page 171.
UseSourceNumberAsDisplayN
ame
For a description of this parameter, refer to ''Routing General
Parameters'' on page 171.
UseDisplayNameAsSourceNum
ber
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
AlwaysUseRouteTable
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
Prefix
This ini file table parameter configures the 'Tel to IP Routing'
table for routing Tel-to-IP calls and the 'Outbound IP Routing'
table for IP-to-IP calls. The format of this parameter is as follows:
[PREFIX]
FORMAT PREFIX_Index = PREFIX_DestinationPrefix,
PREFIX_DestAddress, PREFIX_SourcePrefix,
PREFIX_ProfileId, PREFIX_MeteringCode, PREFIX_DestPort,
PREFIX_SrcIPGroupID, PREFIX_DestHostPrefix,
PREFIX_DestIPGroupID, PREFIX_SrcHostPrefix,
PREFIX_TransportType, PREFIX_SrcTrunkGroupID;
[\PREFIX]
For example:
[PREFIX]
FORMAT PREFIX_Index = PREFIX_DestinationPrefix,
PREFIX_DestAddress, PREFIX_SourcePrefix,
PREFIX_ProfileId, PREFIX_MeteringCode, PREFIX_DestPort,
PREFIX_SrcIPGroupID, PREFIX_DestHostPrefix,
PREFIX_DestIPGroupID, PREFIX_SrcHostPrefix,
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Parameter
Description
PREFIX_TransportType, PREFIX_SrcTrunkGroupID;
PREFIX 0 = *, quest, *, 0, 255, $$, -1, , 1, , -1, -1;
PREFIX 1 = 20, 10.33.37.77, *, 0, 255, $$, -1, , 2, , 0, -1;
PREFIX 2 = 30, 10.33.37.79, *, 1, 255, $$, -1, , -1, , 2, -1;
[\PREFIX]
Notes:
PSTNPrefix
ƒ
This parameter can include up to 50 indices.
ƒ
For a description of these parameters, refer to the
corresponding Web parameters in ''Tel to IP Routing Table''
on page 175 or Outbound IP Routing Table on page 178.
ƒ
The parameter PREFIX_MeteringCode is not applicable.
ƒ
The destination and source phone prefixes
(PREFIX_DestinationPrefix and PREFIX_SourcePrefix
respectively) can be a single number or a range of numbers.
ƒ
Parameters can be skipped using two dollar ($$) symbols, for
example:
Prefix = $$,10.2.10.2,202,1.
ƒ
The destination IP address (PREFIX_DestAddress) can be
either in dotted-decimal notation or FQDN. If an FQDN is
used, DNS resolution is performed according to
DNSQueryType.
ƒ
If the string 'ENUM' is specified for the destination IP address,
an ENUM query containing the destination phone number is
sent to the DNS server. The ENUM reply includes a SIP URI
used as the Request-URI in the outgoing INVITE and for
routing (if Proxy is not used).
ƒ
The IP address can include wildcards. The 'x' wildcard is used
to represent single digits, e.g., 10.8.8.xx represents all
addresses between 10.8.8.10 to 10.8.8.99. The '*' wildcard
represents any number between 0 and 255, e.g., 10.8.8.*
represents all addresses between 10.8.8.0 and 10.8.8.255.
ƒ
For available notations, refer to ''Dialing Plan Notation'' on
page 168.
ƒ
For a description on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
This ini file table parameter configures the routing of IP-to-Tel
calls to Trunk Groups or Inbound IP Routing for IP-to-IP calls.
The format of this parameter is as follows:
[PSTNPrefix]
FORMAT PstnPrefix_Index = PstnPrefix_DestPrefix,
PstnPrefix_TrunkGroupId, PstnPrefix_SourcePrefix,
PstnPrefix_SourceAddress, PstnPrefix_ProfileId,
PstnPrefix_SrcIPGroupID, PstnPrefix_DestHostPrefix,
PstnPrefix_SrcHostPrefix;
[\PSTNPrefix]
For example:
[PSTNPrefix]
FORMAT PstnPrefix_Index = PstnPrefix_DestPrefix,
PstnPrefix_TrunkGroupId, PstnPrefix_SourcePrefix,
PstnPrefix_SourceAddress, PstnPrefix_ProfileId,
PstnPrefix_SrcIPGroupID, PstnPrefix_DestHostPrefix,
PstnPrefix_SrcHostPrefix;
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Parameter
Description
PstnPrefix 0 = 100, 1, 200, *, 0, 2, , ;
PstnPrefix 1 = *, 2, *, , 1, 3, acl, joe;
[\PSTNPrefix]
Notes:
ƒ
This parameter can include up to 24 indices.
ƒ
For a description of these parameters, refer to the
corresponding Web parameters in ''IP to Trunk Group
Routing Table'' on page 181 or Inbound IP Routing Table on
page 184 (for IP-to-IP calls).
ƒ
To support the In-Call Alternative Routing feature, you can
use two entries that support the same call, but assigned with
a different Trunk Group. The second entry functions as an
alternative selection if the first rule fails as a result of one of
the release reasons listed in the AltRouteCauseIP2Tel table.
ƒ
Selection of Trunk Groups (for IP-to-Tel calls) is according to
destination number, source number,and source IP address.
ƒ
The source IP address (SourceAddress) can include the 'x'
wildcard to represent single digits. For example: 10.8.8.xx
represents all IP addresses between 10.8.8.10 and 10.8.8.99.
ƒ
The source IP address (SourceAddress) can include the
asterisk ('*') wildcard to represent any number between 0 and
255. For example, 10.8.8.* represents all addresses between
10.8.8.0 and 10.8.8.255.
ƒ
If the source IP address (SourceAddress) includes an FQDN,
DNS resolution is performed according to DNSQueryType.
ƒ
For available notations that represent multiple numbers, refer
to ''Dialing Plan Notation'' on page 168.
ƒ
For a description on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
RemovePrefix
For a description of this parameter, refer to ''Routing General
Parameters'' on page 171.
RouteModeIP2Tel
For a description of this parameter, refer to ''IP to Trunk Group
Routing'' on page 181.
RouteModeTel2IP
For a description of this parameter, refer to ''Tel to IP Routing
Table'' on page 175.
SwapRedirectNumber
For a description of this parameter, refer to “Configuring the
Digital Gateway Parameters” on page 207.
Prefix2RedirectNumber
For a description of this parameter, refer to “Configuring the
Digital Gateway Parameters” on page 207.
SourceManipulationMode
Determines the SIP headers containing the source number after
manipulation:
Version 5.6
ƒ
[0] = Both SIP From and P-Asserted-Id headers contain the
source number after manipulation (default).
ƒ
[1] = Only SIP From header contains the source number after
manipulation, while the P-Asserted-Id header contains the
source number before manipulation.
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Parameter
SwapTel2IPCalled&CallingNum
bers
Description
If enabled, the device swaps the calling and called numbers
received from the Tel side. The INVITE message contains the
swapped numbers. Applicable for Tel-to-IP calls.
ƒ
[0] = Disabled (default)
ƒ
[1] = Swap calling and called numbers
AddTON2RPI
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
NumberMapTel2IP
This ini file table parameter manipulates the destination number
of Tel-to-IP calls. The format of this parameter is as follows:
[NumberMapTel2Ip]
FORMAT NumberMapTel2Ip_Index =
NumberMapTel2Ip_DestinationPrefix,
NumberMapTel2Ip_SourcePrefix,
NumberMapTel2Ip_SourceAddress,
NumberMapTel2Ip_NumberType,
NumberMapTel2Ip_NumberPlan,
NumberMapTel2Ip_RemoveFromLeft,
NumberMapTel2Ip_RemoveFromRight,
NumberMapTel2Ip_LeaveFromRight,
NumberMapTel2Ip_Prefix2Add,
NumberMapTel2Ip_Suffix2Add,
NumberMapTel2Ip_IsPresentationRestricted,
NumberMapTel2Ip_SrcTrunkGroupID, NumberMapTel2Ip_
SrcIPGroupID;
[\NumberMapTel2Ip]
For example:
[NumberMapTel2Ip]
NumberMapTel2Ip 0 = 01,$$,*,0,0,2,$$,$$,971,$$,$$,$$,$$;
NumberMapTel2Ip 1 = 10,10,*,255,255,3,0,5,100,$$,255,$$,$$;
[\NumberMapTel2Ip]
Notes:
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ƒ
This table parameter can include up to 100 indices.
ƒ
The parameters SourceAddress and IsPresentationRestricted
are not applicable. Set these to $$.
ƒ
The parameter RemoveFromLeft, RemoveFromRight,
Prefix2Add, Suffix2Add, LeaveFromRight, NumberType, and
NumberPlan are applied if the called and calling numbers
match the DestinationPrefix and SourcePrefix conditions.
ƒ
The manipulation rules are executed in the following order:
RemoveFromLeft, RemoveFromRight, LeaveFromRight,
Prefix2Add, and Suffix2Add.
ƒ
Parameters can be skipped by using two dollar signs ('$$').
ƒ
Number Plan and Type can optionally be used in Remote
Party ID (RPID) header by using the EnableRPIHeader and
AddTON2RPI parameters.
ƒ
To configure manipulation of destination numbers for Tel-toIP calls using the Web interface, refer to ''Configuring the
Number Manipulation Tables'' on page 164).
ƒ
For a description on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
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Parameter
NumberMapIP2Tel
Description
This ini file table parameter manipulates the destination number
of IP-to-Tel calls. The format of this parameter is as follows:
[NumberMapIp2Tel]
FORMAT NumberMapIp2Tel_Index =
NumberMapIp2Tel_DestinationPrefix,
NumberMapIp2Tel_SourcePrefix,
NumberMapIp2Tel_SourceAddress,
NumberMapIp2Tel_NumberType,
NumberMapIp2Tel_NumberPlan,
NumberMapIp2Tel_RemoveFromLeft,
NumberMapIp2Tel_RemoveFromRight,
NumberMapIp2Tel_LeaveFromRight,
NumberMapIp2Tel_Prefix2Add,
NumberMapIp2Tel_Suffix2Add,
NumberMapIp2Tel_IsPresentationRestricted;
[\NumberMapIp2Tel]
For example:
[NumberMapIp2Tel]
NumberMapIp2Tel 0 = 01,034,10.13.77.8,$$,0,$$,2,$$,667,$$;
NumberMapIp2Tel 1 = 10,10,1.1.1.1,255,255,3,0,5,100,$$,255;
[\NumberMapIp2Tel]
Notes:
Version 5.6
ƒ
This table parameter can include up to 100 indices.
ƒ
The parameter NumberMapIp2Tel_IsPresentationRestricted
is not applicable. Set its value to $$.
ƒ
RemoveFromLeft, RemoveFromRight, Prefix2Add,
Suffix2Add, LeaveFromRight, NumberType, and NumberPlan
are applied if the called and calling numbers match the
DestinationPrefix, SourcePrefix, and SourceAddress
conditions.
ƒ
The manipulation rules are executed in the following order:
RemoveFromLeft, RemoveFromRight, LeaveFromRight,
Prefix2Add, and Suffix2Add.
ƒ
Parameters can be skipped using two dollar signs ('$$').
ƒ
The Source IP address can include the 'x' wildcard to
represent single digits. For example: 10.8.8.xx represents all
addresses between 10.8.8.10 and 10.8.8.99.
ƒ
The Source IP address can include the asterisk ('*') wildcard
to represent any number between 0 and 255. For example,
10.8.8.* represents all the addresses between 10.8.8.0 and
10.8.8.255.
ƒ
To configure manipulation of destination numbers for IP-toTel calls using the Web interface, refer to ''Configuring the
Number Manipulation Tables'' on page 164).
ƒ
For a description on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
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Parameter
SourceNumberMapTel2IP
Description
This ini file table parameter manipulates the source phone
number for Tel-to-IP calls. The format of this parameter is as
follows:
[SourceNumberMapTel2Ip]
FORMAT SourceNumberMapTel2Ip_Index =
SourceNumberMapTel2Ip_DestinationPrefix,
SourceNumberMapTel2Ip_SourcePrefix,
SourceNumberMapTel2Ip_SourceAddress,
SourceNumberMapTel2Ip_NumberType,
SourceNumberMapTel2Ip_NumberPlan,
SourceNumberMapTel2Ip_RemoveFromLeft,
SourceNumberMapTel2Ip_RemoveFromRight,
SourceNumberMapTel2Ip_LeaveFromRight,
SourceNumberMapTel2Ip_Prefix2Add,
SourceNumberMapTel2Ip_Suffix2Add,
SourceNumberMapTel2Ip_IsPresentationRestricted,
NumberMapTel2Ip_SrcTrunkGroupID,
NumberMapTel2Ip_SrcIPGroupID;
[\SourceNumberMapTel2Ip]
For example:
[SourceNumberMapTel2Ip]
SourceNumberMapTel2Ip 0 =
22,03,$$,0,0,$$,2,$$,667,$$,0,$$,$$;
SourceNumberMapTel2Ip 0 =
10,10,*,255,255,3,0,5,100,$$,255,$$,$$;
[\SourceNumberMapTel2Ip]
Notes:
SourceNumberMapIP2Tel
ƒ
This table parameter can include up to 120 indices.
ƒ
RemoveFromLeft, RemoveFromRight, Prefix2Add,
Suffix2Add, LeaveFromRight, NumberType, NumberPlan,
and IsPresentationRestricted are applied if the called and
calling numbers match the DestinationPrefix and
SourcePrefix conditions.
ƒ
The manipulation rules are executed in the following order:
RemoveFromLeft, RemoveFromRight, LeaveFromRight,
Prefix2Add, and Suffix2Add.
ƒ
Parameters can be skipped by using two dollar signs ('$$').
ƒ
An asterisk ('*') represents all IP addresses.
ƒ
IsPresentationRestricted is set to 'Restricted' only if 'Asserted
Identity Mode' is set to 'P-Asserted'.
ƒ
Number Plan and Type can optionally be used in Remote
Party ID (RPID) header by using the EnableRPIHeader and
AddTON2RPI parameters.
ƒ
To configure manipulation of source numbers for Tel-to-IP
calls using the Web interface, refer to ''Configuring the
Number Manipulation Tables'' on page 164).
ƒ
For a description on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
This ini file table parameter manipulates the source number for
IP-to-Tel calls. The format of this parameter is as follows:
[SourceNumberMapIp2Tel]
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Parameter
Description
FORMAT SourceNumberMapIp2Tel_Index =
SourceNumberMapIp2Tel_DestinationPrefix,
SourceNumberMapIp2Tel_SourcePrefix,
SourceNumberMapIp2Tel_SourceAddress,
SourceNumberMapIp2Tel_NumberType,
SourceNumberMapIp2Tel_NumberPlan,
SourceNumberMapIp2Tel_RemoveFromLeft,
SourceNumberMapIp2Tel_RemoveFromRight,
SourceNumberMapIp2Tel_LeaveFromRight,
SourceNumberMapIp2Tel_Prefix2Add,
SourceNumberMapIp2Tel_Suffix2Add,
SourceNumberMapIp2Tel_IsPresentationRestricted;
[\SourceNumberMapIp2Tel]
For example:
[SourceNumberMapIp2Tel]
SourceNumberMapIp2Tel 0 = 22,03,$$,$$,$$,$$,2,667,$$,$$;
SourceNumberMapIp2Tel 1 =
034,01,1.1.1.1,$$,0,2,$$,$$,972,$$,10;
[\SourceNumberMapIp2Tel]
Notes:
ƒ
RemoveFromLeft, RemoveFromRight, Prefix2Add,
Suffix2Add, LeaveFromRight, NumberType, and NumberPlan
are applied if the called and calling numbers match the
DestinationPrefix, SourcePrefix, and SourceAddress
conditions.
ƒ
The manipulation rules are executed in the following order:
RemoveFromLeft, RemoveFromRight, LeaveFromRight,
Prefix2Add, and Suffix2Add.
ƒ
Parameters can be skipped by using two dollar signs ('$$').
ƒ
The Source IP address can include the 'x' wildcard to
represent single digits. For example: 10.8.8.xx represents all
addresses between 10.8.8.10 and 10.8.8.99.
ƒ
The Source IP address can include the asterisk ('*') wildcard
to represent any number between 0 and 255. For example,
10.8.8.* represents all the addresses between 10.8.8.0 and
10.8.8.255.
ƒ
To configure manipulation of source numbers for IP-to-Tel
calls using the Web interface, refer to ''Configuring the
Number Manipulation Tables'' on page 164).
ƒ
For a description on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
For ETSI ISDN variant, the following Number Plan and Type combinations (Plan/Type) are supported
in the Destination and Source Manipulation tables:
ƒ
0,0 = Unknown, Unknown
ƒ
9,0 = Private, Unknown
ƒ
9,1 = Private, Level 2 Regional
ƒ
9,2 = Private, Level 1 Regional
ƒ
9,3 = Private, PISN Specific
ƒ
9,4 = Private, Level 0 Regional (local)
ƒ
1,0 = Public(ISDN/E.164), Unknown
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Parameter
Description
ƒ
1,1 = Public(ISDN/E.164), International
ƒ
1,2 = Public(ISDN/E.164), National
ƒ
1,3 = Public(ISDN/E.164), Network Specific
ƒ
1,4 = Public(ISDN/E.164), Subscriber
ƒ
1,6 = Public(ISDN/E.164), Abbreviated
For NI-2 and DMS-100 ISDN variants the valid combinations of TON and NPI for calling and called
numbers are (Plan/Type):
ƒ
0/0 - Unknown/Unknown
ƒ
1/1 - International number in ISDN/Telephony numbering plan
ƒ
1/2 - National number in ISDN/Telephony numbering plan
ƒ
1/4 - Subscriber (local) number in ISDN/Telephony numbering plan
ƒ
9/4 - Subscriber (local) number in Private numbering plan
SecureCallsFromIP
For a description of this parameter, refer to ''Advanced
Parameters'' on page 151.
AltRouteCauseTel2IP
This ini file table parameter configures SIP call failure reason
values received from the IP side. If a call is released as a result
of one of these reasons, the device attempts to locate an
alternative route to the call in the 'Tel to IP Routing' table (if
Proxy is not used) or used as a redundant Proxy (when Proxy is
used).
The format of this parameter is as follows:
[AltRouteCauseTel2IP]
FORMAT AltRouteCauseTel2IP_Index =
AltRouteCauseTel2IP_ReleaseCause;
[\AltRouteCauseTel2IP]
For example:
[AltRouteCauseTel2IP]
AltRouteCauseTel2IP 0 = 486; (Busy Here)
AltRouteCauseTel2IP 1 = 480; (Temporarily Unavailable)
AltRouteCauseTel2IP 2 = 408; (No Response)
[\AltRouteCauseTel2IP]
Notes:
AltRouteCauseIP2Tel
ƒ
The 408 reason can be used to specify no response from the
remote party to the INVITE request.
ƒ
This parameter can include up to 5 indices.
ƒ
For defining the Reasons for Alternative Routing table using
the Web interface, refer to ''Reasons for Alternative Routing''
on page 188.
ƒ
For an explanation on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
This ini file table parameter configures call failure reason values
received from the PSTN side (in Q.931 presentation). If a call is
released as a result of one of these reasons, the device attempts
to locate an alternative Trunk Group for the call in the 'IP to
Trunk Group Routing' table.
The format of this parameter is as follows:
[AltRouteCauseIP2Tel]
FORMAT AltRouteCauseIP2Tel_Index =
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Parameter
Description
AltRouteCauseIP2Tel_ReleaseCause;
[\AltRouteCauseIP2Tel]
For example:
[AltRouteCauseIP2Tel]
AltRouteCauseIP2Tel 0 = 3 (No Route to Destination)
AltRouteCauseIP2Tel 1 = 1 (Unallocated Number)
AltRouteCauseIP2Tel 2 = 17 (Busy Here)
[\AltRouteCauseIP2Tel]
Notes:
EnableETSIDiversion
ƒ
This parameter can include up to 5 indices.
ƒ
If the device fails to establish a call to the PSTN because it
has no available channels in a specific trunk group (e.g., all
trunk group's channels are occupied, or the trunk group's
spans are disconnected or out of sync), it uses the Internal
Release Cause '3' (No Route to Destination). This cause can
be used in the AltRouteCauseIP2Tel table to define routing to
an alternative trunk group.
ƒ
For defining the Reasons for Alternative Routing table using
the Web interface, refer to ''Reasons for Alternative Routing''
on page 188.
ƒ
For an explanation on usng ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
Defines the method in which the Redirect Number is passed
toward the Tel side.
ƒ
[0] = Q.931 Redirecting Number Information Element (IE)
(default)
ƒ
[1] = ETSI DivertingLegInformation2 in a Facility IE
CopyDest2RedirectNumber
For a description of this parameter, refer to “Configuring the
Digital Gateway Parameters” on page 207.
FilterCalls2IP
For a description of this parameter, refer to ''Advanced
Parameters'' on page 151.
Alternative Routing Parameters
RedundantRoutingMode
For a description of this parameter, refer to ''Proxy & Registration
Parameters'' on page 132.
AltRoutingTel2IPEnable
For a description of this parameter, refer to ''Routing General
Parameters'' on page 171.
AltRoutingTel2IPMode
For a description of this parameter, refer to ''Routing General
Parameters'' on page 171.
AltRoutingTel2IPConnMethod
For a description of this parameter, refer to ''Routing General
Parameters'' on page 171.
AltRoutingTel2IPKeepAliveTim
e
For a description of this parameter, refer to ''Routing General
Parameters'' on page 171.
IPConnQoSMaxAllowedPL
For a description of this parameter, refer to ''Routing General
Parameters'' on page 171.
IPConnQoSMaxAllowedDelay
For a description of this parameter, refer to ''Routing General
Parameters'' on page 171.
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Parameter
Description
Phone-Context Parameters
AddPhoneContextAsPrefix
For a description of this parameter, refer to ''Mapping NPI/TON
to Phone-Context'' on page 170.
PhoneContext
This ini file table parameter defines the Phone Context table. The
format for this parameter is as follows:
[PhoneContext]
FORMAT PhoneContext_Index = PhoneContext_Npi,
PhoneContext_Ton, PhoneContext_Context;
[\PhoneContext]
Where,
ƒ
Npi = Number Plan.
ƒ
Ton = Type of Number.
ƒ
Context = Phone-Context value.
When a call is received from the ISDN, the NPI and TON are
compared to the table, and the Phone-Context value is used in
the outgoing SIP INVITE message. The same mapping occurs
when an INVITE with a Phone-Context attribute is received. The
Phone-Context parameter appears in the standard SIP headers
where a phone number is used (Request-URI, To, From,
Diversion).
For example:
[PhoneContext]
PhoneContext 0 = 0,0,unknown.com
PhoneContext 1 = 1,1,host.com
PhoneContext 2 = 9,1,na.e164.host.com
[\PhoneContext]
Notes:
SIP User's Manual
ƒ
This parameter can include up to 20 indices.
ƒ
Several entries with the same NPI-TON or Phone-Context are
allowed. In this scenario, a Tel-to-IP call uses the first match.
ƒ
Phone-Context '+' is a unique as it doesn't appear in the
Request-URI as a Phone-Context parameter. Instead, it's
added as a prefix to the phone number. The '+' isn't removed
from the phone number in the IP-to-Tel direction.
ƒ
To configure the Phone Context table using the Web
interface, refer to ''Mapping NPI/TON to Phone-Context'' on
page 170.
ƒ
For a description on using ini file table parameters, refer to
''Structure of ini File Table Parameters'' on page 257.
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4. ini File Configuration
4.4.13 Channel Parameters
The channel-related ini file configuration parameters are described in the table below. The
channel parameters define the DTMF, fax and modem transfer modes.
Table 4-13: Channel ini File Parameters
Parameter
Description
DJBufMinDelay
For a description of this parameter, refer to ''Configuring the RTP /
RTCP Settings'' on page 71.
DJBufOptFactor
For a description of this parameter, refer to ''Configuring the RTP /
RTCP Settings'' on page 71.
FaxTransportMode
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
FaxRelayEnhancedRedunda
ncyDepth
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
FaxRelayRedundancyDepth
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
FaxRelayMaxRate
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
FaxRelayECMEnable
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
FaxModemBypassCoderType
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
CNGDetectorMode
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
FaxCNGMode
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
FaxModemBypassM
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
FaxModemNTEMode
Determines whether the device sends RFC 2833 ANS/ANSam
events upon detection of fax and/or modem answer tones (i.e.,
CED tone).
ƒ
[0] = Disabled (default).
ƒ
[1] = Enabled.
Note: This parameter is applicable only when the fax or modem
transport type is set to bypass or Transparent-with-Events.
FaxBypassPayloadType
For a description of this parameter, refer to ''Configuring the RTP /
RTCP Settings'' on page 71.
CallerIDTransportType
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
ModemBypassPayloadType
Modem Bypass dynamic payload type.
The range is 0-127. The default value is 103.
FaxModemRelayVolume
Determines the fax gain control.
The range -18 to -3 corresponds to -18 dBm to -3 dBm in 1-dB
steps. The default is -6 dBm fax gain control.
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Parameter
Description
FaxBypassOutputGain
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
ModemBypassOutputGain
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
T38MaxDatagram
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
T38FaxMaxBufferSize
Defines the maximum size (in bytes) of a T.38 buffer supported by
the device. This value is included in the outgoing SDP when T.38 is
used for fax relay over IP.
The valid range is 100 to 1,024. The default value is 1,024.
DetFaxOnAnswerTone
For a description of this parameter, refer to ''SIP General
Parameters'' on page 121.
NTEMaxDuration
Maximum time for sending Named Telephony Events (NTEs) to the
IP side, regardless of the time range when the TDM signal is
detected.
The range is -1 to 200,000,000 msec (i.e., 55 hours). The default is
-1 (i.e., NTE stops only upon detection of an End event).
EchoCancellerAggressiveNL
P
Enables or disables the Aggressive NLP at the first 0.5 second of
the call. When enabled, the echo is removed only in the first half a
second of the incoming IP signal.
FaxModemBypassBasicRTP
PacketInterval
ƒ
[0] = Disable (default)
ƒ
[1] = Enable
Determines the basic frame size that is used during fax / modem
bypass sessions.
ƒ
[0] = Determined internally (default)
ƒ
[1] = 5 msec (not recommended)
ƒ
[2] = 10 msec
ƒ
[3] = 20 msec
Note: When set for 5 msec (1), the maximum number of
simultaneous channels supported is 120.
FaxModemBypassDJBufMin
Delay
Determines the Jitter Buffer delay (in milliseconds) during fax and
modem bypass session.
The range is 0 to 150 msec. The default is 40.
EnableFaxModemInbandNet
workDetection
Enables or disables in-band network detection related to
fax/modem.
ƒ
[0] = Disable (default)
ƒ
[1] = Enable
When this parameter is enabled on Bypass mode
(VxxTransportType = 2), a detection of an Answer Tone from the
network triggers a switch to bypass mode in addition to the local
Fax/Modem tone detections. However, only a high bit-rate coder
voice session effectively detects the Answer Tone sent by a remote
Endpoint. This can be useful when, for example, the payload of
voice and bypass is the same, allowing the originator to switch to
bypass mode as well.
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4. ini File Configuration
Parameter
NSEMode
Description
Cisco compatible fax and modem bypass mode.
ƒ
[0] = NSE disabled (default)
ƒ
[1] = NSE enabled
Notes:
ƒ
This feature can be used only if VxxModemTransportType = 2
(Bypass).
ƒ
If NSE mode is enabled, the SDP contains the following line:
'a=rtpmap:100 X-NSE/8000'.
ƒ
To use this feature:
-- The Cisco gateway must include the following definition:
'modem passthrough nse payload-type 100 codec g711alaw'.
-- Set the Modem transport type to Bypass mode
(VxxModemTransportType = 2) for all modems.
-- Configure the gateway parameter NSEPayloadType = 100.
In NSE bypass mode, the device starts using G.711 A-Law
(default) or G.711μ-Law according to the parameter
FaxModemBypassCoderType. The payload type used with these
G.711 coders is a standard one (8 for G.711 A-Law and 0 for
G.711 μ-Law). The parameters defining payload type for the 'old'
AudioCodes' Bypass mode FaxBypassPayloadType and
ModemBypassPayloadType are not used with NSE Bypass. The
bypass packet interval is selected according to the parameter
FaxModemBypassBasicRtpPacketInterval.
NSEPayloadType
NSE payload type for Cisco Bypass compatible mode.
The valid range is 96-127. The default value is 105.
Note: Cisco gateways usually use NSE payload type of 100.
V21ModemTransportType
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
V22ModemTransportType
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
V23ModemTransportType
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
V32ModemTransportType
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
V34ModemTransportType
For a description of this parameter, refer to ''Configuring the Fax /
Modem / CID Settings'' on page 67.
V34FaxTransportType
Determines the V.34 fax transport method.
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ƒ
[0] = Transparent
ƒ
[1] = Relay (default)
ƒ
[2] = Bypass
ƒ
[3] = Transparent with Events
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Parameter
UserDefinedToneDetectorEn
able
BellModemTransportType
Description
Enables or disables detection of User Defined Tones signaling.
ƒ
[0] = Disable (default)
ƒ
[1] = Enable
Determines the Bell modem transport method.
ƒ
[0] = Transparent (default).
ƒ
[2] = Bypass.
ƒ
[3] = Transparent with events.
InputGain
For a description of this parameter, refer to ''Configuring the Voice
Settings'' on page 66.
VoiceVolume
For a description of this parameter, refer to ''Configuring the Voice
Settings'' on page 66.
RTPRedundancyDepth
For a description of this parameter, refer to ''Configuring the RTP /
RTCP Settings'' on page 71.
RFC2198PayloadType
For a description of this parameter, refer to ''Configuring the RTP /
RTCP Settings'' on page 71.
EnableSilenceCompression
For a description of this parameter, refer to ''Configuring the Voice
Settings'' on page 66.
IsCiscoSCEMode
Determines whether a Cisco gateway exists at the remote side.
ƒ
[0] = No Cisco gateway exists at the remote side (default).
ƒ
[1] = A Cisco gateway exists at the remote side.
When there is a Cisco gateway at the remote side, the device must
set the value of the 'annexb' parameter of the fmtp attribute in the
SDP to 'no'. This logic is used if EnableSilenceCompression = 2
(enable without adaptation). In this case, Silence Suppression is
used on the channel, but not declared in the SDP.
Note: The IsCiscoSCEMode parameter is only relevant when the
selected coder is G.729.
EnableEchoCanceller
For a description of this parameter, refer to ''Configuring the Voice
Settings'' on page 66.
MaxEchoCancellerLength
For a description of this parameter, refer to “Configuring the
General Media Settings” on page 78.
ECNLPMode
Defines the echo cancellation Non-Linear Processing (NLP) mode.
EchoCancellerAggressiveNL
P
EnableNoiseReduction
ƒ
[0] = NLP adapts according to echo changes (default).
ƒ
[1] = Disables NLP.
Enables or disables the Aggressive Non-Linear Processor (NLP) in
the first 0.5 second of the call.
ƒ
[0] = Disabled (default)
ƒ
[1] = Enabled
Enables / disables the DSP Noise Reduction mechanism.
ƒ
[0] = Disable (default).
ƒ
[1] = Enable.
Note: When this parameter is enabled the channel capacity might
be reduced.
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Parameter
Description
EnableStandardSIDPayloadT
ype
For a description of this parameter, refer to ''Configuring the RTP /
RTCP Settings'' on page 71.
ComfortNoiseNegotiation
For a description of this parameter, refer to ''Configuring the RTP /
RTCP Settings'' on page 71.
RTPSIDCoeffNum
Determines the number of spectral coefficients added to an SID
packet being sent according to RFC 3389. Valid only if
EnableStandardSIDPayloadType is set to 1.
The valid values are [0] (default), [4], [6], [8] and [10].
DTMFVolume
For a description of this parameter, refer to ''Configuring the Voice
Settings'' on page 66.
DTMFGenerationTwist
For a description of this parameter, refer to ''Configuring the Voice
Settings'' on page 66.
DTMFInterDigitInterval
Time in msec between generated DTMF digits to PSTN side (if
TxDTMFOption = 1, 2 or 3).
The default value is 100 msec. The valid range is 0 to 32767.
DTMFDigitLength
Time (in msec) for generating DTMF tones to the PSTN side (if
TxDTMFOption = 1, 2 or 3). It also configures the duration that is
sent in INFO (Cisco) messages.
The valid range is 0 to 32767. The default value is 100.
RxDTMFHangOverTime
Defines the Voice Silence time (in msec units) after playing DTMF
or MF digits to the Tel / PSTN side that arrive as Relay from the IP
side.
Valid range is 0 to 2,000 msec. The default is 1,000 msec.
TxDTMFHangOverTime
Defines the Voice Silence time (in msec) after detecting the end of
DTMF or MF digits at the Tel / PSTN side when the DTMF
Transport Type is either Relay or Mute.
Valid range is 0 to 2,000 msec. The default is 1,000 msec.
DTMFTransportType
For a description of this parameter, refer to ''Configuring the Voice
Settings'' on page 66.
RFC2833PayloadType
For a description of this parameter, refer to ''DTMF & Dialing
Parameters'' on page 147.
R1DetectionStandard
Determines the R1 MF protocol used for detection.
UserDefinedToneDetectorEn
able
ƒ
[0] = ITU (default)
ƒ
[1] = R1.5
Enables or disables detection of User Defined Tones signaling.
ƒ
[0] = Disable
ƒ
[1] = Enable
UDTDetectorFrequencyDevia
tion
Defines the deviation (in Hz) allowed for the detection of each
signal frequency.
The valid range is 1 to 50. The default value is 50.
CPTDetectorFrequencyDevia
tion
Defines the deviation (in Hz) allowed for the detection of each CPT
signal frequency.
The valid range is 1 to 30. The default value is 10.
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Parameter
MGCPDTMFDetectionPoint
KeyBlindTransfer
Description
ƒ
[0] = DTMF event is reported on the end of a detected DTMF
digit.
ƒ
[1] = DTMF event is reported on the start of a detected DTMF
digit (default).
Keypad sequence that activates blind transfer for Tel-to-IP calls.
There are two possible scenarios:
ƒ
Option 1: After this sequence is dialed, the current call is put on
hold (using Re-INVITE), a dial tone is played to the B-channel,
and then phone number collection starts.
ƒ
Option 2: A Hook-Flash is pressed, the current call is put on
hold, a dial tone is played to the B-channel, and then digit
collection starts. After this sequence is identified, the device
continues the collection of the destination phone number.
For both options, after the phone number is collected, it's sent to
the transferee in a SIP REFER request (without a Replaces
header). The call is then terminated and a confirmation tone is
played to the B-channel. If the phone number collection fails due to
a mismatch, a reorder tone is played to the B-channel.
Note: It is possible to configure whether the KeyBlindTransfer code
is added as a prefix to the dialed destination number, by using the
parameter KeyBlindTransferAddPrefix.
KeyBlindTransferAddPrefix
VoicePayloadFormat
Determines whether the device adds the Blind Transfer code
(KeyBlindTransfer) to the dialed destination number.
ƒ
[0] Disable (default).
ƒ
[1] Enable.
Determines the bit ordering of the G.726/G.727 voice payload
format.
ƒ
[0] = Little Endian (default)
ƒ
[1] = Big Endian
Note: To ensure high voice quality when using G.726/G.727, both
communicating ends should use the same endianness format.
Therefore, when the device communicates with a third-party entity
that uses the G.726/G.727 voice coder and voice quality is poor,
change the settings of this parameter (between Big Endian and
Little Endian).
VQMonEnable
For a description of this parameter, refer to “Configuring the RTP /
RTCP Settings” on page 71.
VQMonBurstHR
For a description of this parameter, refer to “Configuring the RTP /
RTCP Settings” on page 71.
VQMonDelayTHR
For a description of this parameter, refer to “Configuring the RTP /
RTCP Settings” on page 71.
VQMonEOCRValTHR
For a description of this parameter, refer to “Configuring the RTP /
RTCP Settings” on page 71.
VQMonGMin
For a description of this parameter, refer to “Configuring the RTP /
RTCP Settings” on page 71.
RTCPInterval
For a description of this parameter, refer to “Configuring the RTP /
RTCP Settings” on page 71.
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Parameter
Description
DisableRTCPRandomize
For a description of this parameter, refer to “Configuring the RTP /
RTCP Settings” on page 71.
RTCPXRESCTransportType
Determines the transport layer used for outgoing SIP dialogs
initiated by the device to the RTCP-XR Collection Server.
ƒ
[-1] Not Configured (default)
ƒ
[0] UDP
ƒ
[1] TCP
ƒ
[2] TLS
Note: When set to ‘Not Configured’, the value of the parameter
SIPTransportType is used.
RTCPXREscIP
For a description of this parameter, refer to “Configuring the RTP /
RTCP Settings” on page 71.
RTCPXRReportMode
For a description of this parameter, refer to “Configuring the RTP /
RTCP Settings” on page 71.
4.4.14 Auxiliary / Configuration Files Parameters
The configuration files (i.e., auxiliary files) can be loaded to the device using the Web
interface or a TFTP session (refer to ''Auxiliary Files'' on page 231). Before you load them
to the device, you need to specify these files in the ini file and whether they must be stored
in the non-volatile memory. The table below lists the ini file parameters associated with
these auxiliary files:
Table 4-14: Auxiliary / Configuration ini File Parameters
Parameter
Description
CallProgressTonesFilename
The name of the file containing the Call Progress Tones
definitions. Refer to the Product Reference Manual for
additional information on how to create and load this file.
CASFileName
This is the name of the file containing specific CAS protocol
definition (such as 'E_M_WinkTable.dat'). These files are
provided to support various types of CAS signaling.
CASFileName_x
CAS file name (e.g., 'E_M_WinkTable.dat') that defines the
CAS protocol. It is possible to define up to eight different CAS
files by repeating this parameter. Each CAS file can be
associated with one or more of the device trunks using the
parameter CASTableIndex_x.
CASTablesNum
Number 1 to 8. Specifies how many CAS configuration files
are loaded.
VoicePromptsFileName
The name (and path) of the file containing the Voice Prompts
definitions. Refer to the Product Reference Manual for
additional information on how to create and load this file.
PrerecordedTonesFileName
The name (and path) of the file containing the Prerecorded
Tones.
CasTrunkDialPlanName
The Dial Plan name (up to 11-character strings) that is used
on the specific trunk.
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Parameter
Description
DialPlanFileName
The name (and path) of the file containing dial-plan
configuration for CAS and SIP protocols. This file should be
constructed using the TrunkPack Conversion Utility (refer to
the Product Reference Manual) supplied as part of the
software package on the CD accompanying the device.
UserInfoFileName
The name (and path) of the file containing the User
Information data.
SetDefaultOnIniFileProcess
Determines if all the device's parameters are set to their
defaults before processing the updated ini file.
SaveConfiguration
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ƒ
[0] Disable - parameters not included in the downloaded ini
file are not returned to default settings (i.e., retain their
current settings).
ƒ
[1] Enable (default)
Determines if the device's configuration (parameters and files)
is saved to flash (non-volatile memory).
ƒ
[0] = Configuration isn't saved to flash memory.
ƒ
[1] = Configuration is saved to flash memory (default).
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5. Default Settings
Default Settings
You can restore the device's factory default settings or define your own default settings for
the device.
5.1
Defining Default Settings
The device is shipped with factory default configuration values stored on its non-volatile
memory (flash). However, you can define your own default values instead of using the
factory defaults. This is performed using an ini file that includes the header
'[ClientDefaults]'. Below this header, simply define new default values for the required ini file
parameters. The parameters are defined in the same format as in the standard ini file, and
loaded to the device using TFTP (i.e., not via the Web interface).
An example of a ClientsDefault ini file for defining default values for Syslog server
parameters is shown below:
[ClientDefaults]
EnableSyslog = 1
SyslogServerIP = 10.13.2.20
¾ To define default values for device parameters, take these 2 steps:
1.
Configure the ClientDefaults ini file with new default parameter values, as required.
2.
Load the ClientDefaults ini file to the device, using TFTP (refer to the Product
Reference Manual).
¾ To remove user-defined defaults and restore factory default values,
take this step:
„
5.2
Load an empty (i.e., without any parameters) ClientDefaults ini file to the device, using
TFTP.
Restoring Factory Defaults
You can restore all or most of the device's configuration settings to default settings:
„
Restoring default settings except for the device's IP address and Web interface's login
user name and password: Load to the device an empty ini file (without any parameters
or with a semicolon (;) preceding all lines). When a parameter is absent from a loaded
ini file, the default value is assigned to that parameter (according to the cmp file loaded
to the device) and saved to the non-volatile memory (thereby, overriding the value
previously defined for that parameter).
„
Restoring all default settings, including the device's IP address and Web interface's
login user name and password: Use the device's hardware Reset button (refer to the
device's Installation Manual).
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6. Auxiliary Configuration Files
Auxiliary Configuration Files
This section describes the auxiliary files (with the dat file extension), which are loaded, in
addition to the ini file, to the device. You can load the auxiliary files to the device using one
of the following methods:
6.1
„
Web interface (refer to ''Loading Auxiliary Files'' on page 231)
„
ini file: specify the name of the relevant auxiliary file in the device's ini file and then
load the ini file to the device (refer to ''Loading Auxiliary Files'' on page 231).
Configuring the Call Progress Tones File
The Call Progress Tones (CPT) auxiliary file used by the device is a binary file (with file
extension dat). This file contains the definitions of the Call Progress Tones (levels and
frequencies) that are detected / generated by the device.
You can either use one of the supplied device auxiliary (dat) files or create your own file. To
create your own auxiliary file, it's recommended to modify the supplied usa_tone.ini file (in
any standard text editor) to suit your specific requirements, and to convert the modified ini
file into binary format using the TrunkPack Downloadable Conversion Utility. For the
description of the procedure on how to convert CPT ini file into a binary dat file, refer to the
Product Reference Manual.
To load the Call Progress Tones (dat) file to the device, use the Web interface or ini file
(refer to ''Loading Auxiliary Files'' on page 231).
Note: Only the dat file can be loaded to the device.
You can create up to 32 different Call Progress Tones, each with frequency and format
attributes. The frequency attribute can be single or dual-frequency (in the range of 300 to
1980 Hz), or an Amplitude Modulated (AM). In total, up to 64 different frequencies are
supported. Only eight AM tones, in the range of 1 to 128 kHz, can be configured (the
detection range is limited to 1 to 50 kHz). Note that when a tone is composed of a single
frequency, the second frequency field must be set to zero.
The format attribute can be one of the following:
„
Continuous: (e.g., dial tone) a steady non-interrupted sound. Only the 'First Signal On
time' should be specified. All other on and off periods must be set to zero. In this case,
the parameter specifies the detection period. For example, if it equals 300, the tone is
detected after 3 seconds (300 x 10 msec). The minimum detection time is 100 msec.
„
Cadence: A repeating sequence of on and off sounds. Up to four different sets of on /
off periods can be specified.
„
Burst: A single sound followed by silence. Only the 'First Signal On time' and 'First
Signal Off time' should be specified. All other on and off periods must be set to zero.
The burst tone is detected after the off time is completed.
You can specify several tones of the same type. These additional tones are used only for
tone detection. Generation of a specific tone conforms to the first definition of the specific
tone. For example, you can define an additional dial tone by appending the second dial
tone's definition lines to the first tone definition in the ini file. The device reports dial tone
detection if either of the two tones is detected.
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The Call Progress Tones section of the ini file comprises the following segments:
„
[NUMBER OF CALL PROGRESS TONES]: Contains the following key:
'Number of Call Progress Tones' defining the number of Call Progress Tones that are
defined in the file.
„
[CALL PROGRESS TONE #X]: containing the Xth tone definition (starting from 1 and
not exceeding the number of Call Progress Tones defined in the first section) using the
following keys:
•
Tone Type: Call Progress Tone types:
♦
[1] Dial Tone
♦
[2] Ringback Tone
♦
[3] Busy Tone
♦
[7] Reorder Tone
♦
[8] Confirmation Tone (Applicable only to Analog devices)
♦
[9] Call Waiting Tone (Applicable only to Analog devices)
♦
[15] Stutter Dial Tone (Applicable only to Analog devices)
♦
[16] Off Hook Warning Tone (Applicable only to Analog devices)
♦
[17] Call Waiting Ringback Tone
♦
[23] Hold Tone
•
Tone Modulation Type: Either Amplitude Modulated (1) or regular (0).
•
Tone Form: The tone's format can be one of the following:
♦
Continuous (1)
♦
Cadence (2)
♦
Burst (3)
•
Low Freq [Hz]: frequency (in Hz) of the lower tone component in case of dual
frequency tone, or the frequency of the tone in case of single tone. This is not
relevant to Amplitude Modulated (AM) tones.
•
High Freq [Hz: frequency (in Hz) of the higher tone component in case of dual
frequency tone, or zero (0) in case of single tone (not relevant to AM tones).
•
Low Freq Level [-dBm]: generation level 0 dBm to -31 dBm in dBm (not relevant
to AM tones).
•
High Freq Level: generation level. 0 to -31 dBm. The value should be set to 32 in
the case of a single tone (not relevant to AM tones).
•
First Signal On Time [10 msec]: 'Signal On' period (in 10 msec units) for the first
cadence on-off cycle. For be continuous tones, this parameter defines the
detection period. For burst tones, it defines the tone's duration.
•
First Signal Off Time [10 msec]: 'Signal Off' period (in 10 msec units) for the first
cadence on-off cycle (for cadence tones). For burst tones, this parameter defines
the off time required after the burst tone ends and the tone detection is reported.
For continuous tones, this parameter is ignored.
•
Second Signal On Time [10 msec]: 'Signal On' period (in 10 msec units) for the
second cadence on-off cycle. Can be omitted if there isn't a second cadence.
•
Second Signal Off Time [10 msec]: 'Signal Off' period (in 10 msec units) for the
second cadence on-off cycle. Can be omitted if there isn't a second cadence.
•
Third Signal On Time [10 msec]: 'Signal On' period (in 10 msec units) for the
third cadence ON-OFF cycle. Can be omitted if there isn't a third cadence.
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6. Auxiliary Configuration Files
•
Third Signal Off Time [10 msec]: 'Signal Off' period (in 10 msec units) for the
third cadence ON-OFF cycle. Can be omitted if there isn't a third cadence.
•
Fourth Signal On Time [10 msec]: 'Signal On' period (in 10 msec units) for the
fourth cadence ON-OFF cycle. Can be omitted if there isn't a fourth cadence.
•
Fourth Signal Off Time [10 msec]: 'Signal Off' period (in 10 msec units) for the
fourth cadence ON-OFF cycle. Can be omitted if there isn't a fourth cadence.
•
Carrier Freq [Hz]: frequency of the carrier signal for AM tones.
•
Modulation Freq [Hz]: frequency of the modulated signal for AM tones (valid
range from 1 to 128 Hz).
•
Signal Level [-dBm]: level of the tone for AM tones.
•
AM Factor [steps of 0.02]: amplitude modulation factor (valid range from 1 to 50.
Recommended values from 10 to 25).
Notes:
•
When the same frequency is used for a continuous tone and a cadence
tone, the 'Signal On Time' parameter of the continuous tone must have a
value that is greater than the 'Signal On Time' parameter of the cadence
tone. Otherwise the continuous tone is detected instead of the cadence
tone.
•
The tones frequency should differ by at least 40 Hz from one tone to
other defined tones.
For example, to configure the dial tone to 440 Hz only, enter the following text:
#Dial tone
[CALL PROGRESS TONE #1]
Tone Type=1
Tone Form =1 (continuous)
Low Freq [Hz]=440
High Freq [Hz]=0
Low Freq Level [-dBm]=10 (-10 dBm)
High Freq Level [-dBm]=32 (use 32 only if a single tone is
required)
First Signal On Time [10msec]=300; the dial tone is detected after
3 sec
First Signal Off Time [10msec]=0
Second Signal On Time [10msec]=0
Second Signal Off Time [10msec]=0
6.2
Prerecorded Tones (PRT) File
The Call Progress Tones (CPT) mechanism has several limitations such as a limited
number of predefined tones and a limited number of frequency integrations in one tone. To
overcome these limitations and provide tone generation capability that is more flexible, the
Prerecorded Tones (PRT) file can be used. If a specific prerecorded tone exists in the PRT
file, it takes precedence over the same tone that exists in the CPT file and is played instead
of it.
Note:
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The PRT is a *.dat file containing a set of prerecorded tones that can be played by the
device. Up to 40 tones (totaling approximately 10 minutes) can be stored in a single PRT
file on the device's flash memory. The prerecorded tones are prepared offline using
standard recording utilities (such as CoolEditTM) and combined into a single file using
AudioCodes' TrunkPack Downloadable Conversion utility (refer to the Product Reference
Manual).
The raw data files must be recorded with the following characteristics:
„
Coders: G.711 A-law or G.711 µ-law
„
Rate: 8 kHz
„
Resolution: 8-bit
„
Channels: mono
The generated PRT file can then be loaded to the device using AudioCodes' BootP/TFTP
utility or the Web interface (refer to ''Loading Auxiliary Files'' on page 231).
The prerecorded tones are played repeatedly. This allows you to record only part of the
tone and then play the tone for the full duration. For example, if a tone has a cadence of 2
seconds on and 4 seconds off, the recorded file should contain only these 6 seconds. The
PRT module repeatedly plays this cadence for the configured duration. Similarly, a
continuous tone can be played by repeating only part of it.
6.3
Voice Prompts File
The voice announcement file contains a set of Voice Prompts that can be played by the
device during operation. The voice announcements are prepared offline using standard
recording utilities and combined into a single file using the TrunkPack Downloadable
Conversion Utility. The generated announcement file can then be loaded to the device
using the BootP/TFTP utility (refer to the Product Reference Manual).
If the size of the combined Voice Prompts file is less than 1 MB, it can permanently be
stored on flash memory. Larger files, up to 10 MB, are stored in RAM, and should be
loaded again (using BootP/TFTP utility) after the device is reset.
The Voice Prompts integrated file is a collection of raw voice recordings and / or wav files.
These recordings can be prepared using standard utilities such as CoolEdit, GoldwaveTM
and others. The raw voice recordings must be sampled at 8000 kHz / mono / 8 bit. The wav
files must be recorded with G.711μ-Law/A-Law/Linear.
When the list of recorded files is converted to a single voiceprompts.dat file, every Voice
Prompt is tagged with an ID number, starting with '1'. This ID is used later by the device to
start playing the correct announcement. Up to 1,000 Voice Prompts can be used.
AudioCodes provides a professionally recorded English (U.S.) Voice Prompts file.
¾ To generate and load the Voice Prompts file, take these 3 steps:
1.
Prepare one or more voice files using standard utilities.
2.
Use the TrunkPack Downloadable Conversion Utility to generate the voiceprompts.dat
file from the pre-recorded voice messages (refer to the Product Reference Manual).
3.
Load the voiceprompts.dat file to the device using TFTP (refer to the Product
Reference Manual) or Web interface (refer to ''Loading Auxiliary Files'' on page 231).
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6.4
6. Auxiliary Configuration Files
CAS Protocol Auxiliary Files
The CAS Protocol auxiliary files contain the CAS Protocol definitions that are used for CASterminated trunks. You can either use the supplied files or construct your own files. Up to
eight files can be loaded and different files can be assigned to different trunks. The CAS
files can be loaded to the device using the Web interface or ini file (refer to ''Loading
Auxiliary Files'' on page 231).
Note: All CAS files loaded together must belong to the same Trunk Type (i.e., either
E1 or T1).
6.5
Dial Plan File
The source file for the Dial Plan configuration contains a list of known prefixes (e.g. area
codes and international telephone number patterns) for the PSTN to which the device is
connected. The device uses this information to detect end-of-dialing in certain CAS
configurations where the end-indicator (ST) is not used. The device supports up to 8,000
distinct prefixes in the dial-plan file.
The CasTrunkDialPlanName ini file parameter determines which Dial Plan (in a Dial Plan
file) to use for a specific trunk (refer to ''Configuring the Trunk Settings'' on page 82). The
Dial Plan can be loaded using the Web interface (refer to ''Loading Auxiliary Files'' on page
231).
The following is an example of an ini file that includes these definitions. This ini file is
converted (using the TrunkPack Conversion Utility - refer to the Product Reference Manual)
to a binary file and loaded to the device.
; Example of dial-plan configuration.
; This file contains two dial plans: you may specify which
; one to use in CAS configuration.
[ PLAN1 ]
; Define the area codes 02, 03, 04.
; In these area codes, phone numbers have 7 digits.
02,7
03,7
04,7
; Define the cellular/VoIP area codes 052, 054, 050, and 077.
; In these area codes, phone numbers have 8 digits.
052,8
054,8
050,8
077,8
; Define the international prefixes 00, 012, 014.
; The number following these prefixes may
; be 7 to 14 digits in length.
00,7-14
012,7-14
014,7-14
; Define the emergency number 911.
; No additional digits are expected.
911,0
[ PLAN2 ]
; Define the area codes 02, 03, 04.
; In these area codes, phone numbers have 7 digits.
0[2-4],7
; Operator services starting with a star: *41, *42, *43.
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; No additional digits are expected.
*4[1-3],0
The list must be prepared in a textual ini file with the following syntax:
„
Every line in the file defines a known dialing prefix and the number of digits expected
to follow that prefix. The prefix must be separated from the number of additional digits
by a comma (',').
„
Empty lines are ignored.
„
Lines beginning with a semicolon (';') are ignored.
„
Multiple dial plans may be specified in one file; A name in square brackets on a
separate line indicates the beginning of a new dial plan. Up to eight dial plans can be
defined.
„
Asterisks ('*') and number-signs ('#') can be specified as part of the prefix.
„
Numeric ranges are allowed in the prefix.
„
A numeric range is allowed in the number of additional digits.
Note: The prefixes must not overlap. Attempting to process an overlapping
configuration in the TrunkPack Conversion Utility results in an error message
specifying the problematic line.
6.6
User Information File
The User Information file is a text file that maps PBX extensions, connected to the device,
to global IP numbers. In this context, a global IP phone number (alphanumerical) serves as
a routing identifier for calls in the 'IP World'. The PBX extension uses this mapping to
emulate the behavior of an IP phone.
Note: The mapping mechanism is disabled by default and must be activated using
the parameter EnableUserInfoUsage (refer to ''Advanced Parameters'' on
page 151).
Each line in the file represents a mapping rule of a single PBX extension. Up to 1,000 rules
can be configured. Each line includes five items separated with commas. The items are
described in the table below:
Table 6-1: User Information Items
Item
Description
Maximum Size
(Characters)
PBX extension #
The relevant PBX extension number.
10
Global phone #
The relevant global phone number.
20
Display name
A string that represents the PBX extensions for the
Caller ID.
30
Username
A string that represents the user name for SIP
registration.
40
Password
A string that represents the password for SIP
registration.
20
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An example of a User Information file is shown in the figure below:
Figure 6-1: Example of a User Information File
Note: The last line in the User Information file must end with a carriage return (i.e.,
by pressing the <Enter> key).
The User Information file can be loaded to the device using the ini file (UserInfoFileName
parameter described in ''Auxiliary / Configuration Files Parameters'' on page 331), the Web
interface (refer to ''Loading Auxiliary Files'' on page 231), or by using the automatic update
mechanism (UserInfoFileURL, refer to the Product Reference Manual).
The maximum permissible size of the file is 108,000 bytes.
Each PBX extension registers separately (a REGISTER message is sent for each entry
only if AuthenticationMode is set to Per Endpoint) using the IP number in the From / To
headers. The REGISTER messages are sent gradually. Initially, the device sends requests
according to the maximum number of allowed SIP dialogs (configured by the parameter
NumberOfActiveDialogs). After each received response, the subsequent request is sent.
Therefore, no more than NumberOfActiveDialogs dialogs are active simultaneously. The
user name and password are used for SIP Authentication when required.
The calling number of outgoing Tel-to-IP calls is first translated to an IP number and then (if
defined), the manipulation rules are performed. The Display Name is used in the From
header in addition to the IP number. The called number of incoming IP-to-Tel calls is
translated to a PBX extension only after manipulation rules (if defined) are performed.
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7. IP Telephony Capabilities
IP Telephony Capabilities
This section describes the device's IP telephony capabilities.
7.1
IP-to-IP Routing (SIP Trunking)
The AudioCodes device supports IP-to-IP VoIP call routing (or SIP trunking). The device
enables Enterprises to seamlessly connect their IP-PBX to a SIP trunk provided by an
Internet Telephony Service Provider (ITSP). The Enterprise can communicate with the
PSTN through the ITSP, which interfaces directly with PSTN. Alternatively, the device can
also provide the interface with the PSTN.
At the same time, the device can also provide an interface with the traditional PSTN
network, enabling PSTN fallback in case of IP network failure. In addition, the device
supports multiple SIP trunks, whereby if a connection to one ITSP goes down, the call can
immediately be transferred to another ITSP. By allowing multiple SIP trunks where each
trunk is designated for a specific ITSP, the device can route calls to an ITSP, based on call
destination (e.g., country code).
Therefore, in addition to providing VoIP communication within an Enterpise's LAN, the
device allows the Enterprise to communicate outside of the corporate LAN, using SIP
trunking.
The device interfaces between the Enterprise's IP-PBX and ITSP, allowing SIP trunking
implementation by the Enterprise, for example, in the following scenarios:
„
VoIP between headquarters and remote offices
„
VoIP between Enterprise and PSTN via their ITSP
For a detailed explanation on configuring IP-to-IP call routing, refer to the document IP-to-IP
SIP Call Routing Application Note.
7.2
Answer Machine Detector (AMD)
Answering Machine Detection can be useful in automatic dialing applications. In some of
these applications, it is important to detect if a human voice or answering machine is
answering the call. Answering Machine Detection can be activated and de-activated only
after a channel is already open. The direction of the detection (PSTN or IP) can be
configured (using the parameter AMDDetectionDirection - refer to ''Media Server
Parameters'' on page 300), as well as the detector detection sensitivity using the parameter
AMDDetectionSensitivity - refer to ''Configuring the IPmedia Settings'' on page 76).
Upon every Answering Machine Detection activation, the device can send a SIP INFO
message to an Application server, notifying it of one of the following:
„
Human voice has been detected
„
Answering machine has been detected
„
Silence (i.e., no voice detected) has been detected
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The table below shows the success rates of the AMD feature for correctly detecting live and
fax calls:
Table 7-1: Approximate AMD Detection Sensitivity (Based on North American English)
Performance
AMD Detection
Sensitivity
Success Rate for Live Calls
Success Rate for Answering Machine
0 (Best for
Answering
Machine)
-
-
1
82.56%
97.10%
2
85.87%
96.43%
3 (Default)
88.57%
94.76%
4
88.94%
94.31%
5
90.42%
91.64%
6
90.66%
91.30%
7 (Best for Live
Calls)
94.72%
76.14%
A pre-requisite for enabling the AMD feature is to set the ini file parameter
EnableDSPIPMDetectors to 1. In addition, to enable voice detection, required once the
AMD detects the answering machine, the ini file parameter EnableVoiceDetection must be
set to 1.
Note: The device's AMD feature is based on voice detection for North American
English. If you want to implement AMD in a different language or region, you
must provide AudioCodes with a database of recorded voices in the language
on which the device's AMD mechanism can base its voice detector algorithms
for detecting these voices. The data needed for an accurate calibration should
be recorded under the following guidelines:
SIP User's Manual
•
Statistical accuracy: The number of recordings should be large (i.e.,
about 100) and varied. The calls must be made to different people, at
different times. The calls must be made in the specific location in which
the device's AMD mechanism is to operate.
•
Real-life recording: The recordings should simulate real-life answering of
a person picking up the phone without the caller speaking (until the AMD
decision).
•
Normal environment interferences: The environment should almost
simulate real-life scenarios, i.e., not sterile but not too noisy either.
Interferences, for example, could include background noises of other
people talking, spikes, and car noises.
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The SIP call flows below show an example of implementing the device's AMD feature. This
scenario example allows a third-party Application server to play a recorded voice message
to an answering machine.
1.
Upon detection by the device of the answering machine, the device sends a SIP INFO
message to the Application server:
INFO sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.168.249;branch=z9hG4bKac1566945480
Max-Forwards: 70
From: sut <sip:[email protected]:5060>;tag=1c1505895240
To: sipp <sip:[email protected]:5060>;tag=1
Call-ID: [email protected]
CSeq: 1 INFO
Contact: <sip:[email protected]>
Supported: em,timer,replaces,path,resource-priority
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUB
SCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-IPmedia 260_UN/v.5.20A.040.004
Content-Type: application/x-detect
Content-Length: 30
Type= AMD
SubType= AUTOMATA
2.
The device then detects the start of voice (i.e., the greeting message of the answering
machine), and then sends the following to the Application server:
INFO sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.168.249;branch=z9hG4bKac482466515
Max-Forwards: 70
From: sut <sip:[email protected]:5060>;tag=1c419779142
To: sipp <sip:[email protected]:5060>;tag=1
Call-ID: [email protected]
CSeq: 1 INFO
Contact: <sip:[email protected]>
Supported: em,timer,replaces,path,resource-priority
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUB
SCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-IPmedia 260_UN/v.5.20A.040.004
Content-Type: application/x-detect
Content-Length: 34
Type= PTT
SubType= SPEECH-START
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3.
Upon detection of the end of voice (i.e., end of the greeting message of the answering
machine), the device sends the Application server the following:
INFO sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.168.249;branch=z9hG4bKac482466515
Max-Forwards: 70
From: sut <sip:[email protected]:5060>;tag=1c419779142
To: sipp <sip:[email protected]:5060>;tag=1
Call-ID: [email protected]
CSeq: 1 INFO
Contact: <sip:[email protected]>
Supported: em,timer,replaces,path,resource-priority
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUB
SCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-IPmedia 260_UN/v.5.20A.040.004
Content-Type: application/x-detect
Content-Length: 34
Type= PTT
SubType= SPEECH-END
4.
The Application server now sends its message to the answering message.
If the device detects voice and not an answering machine, the SIP INFO message includes:
Type= AMD
SubType= VOICE
If the device detects silence, the SIP INFO message includes the SubType SILENT.
7.3
Stand-Alone Survivability (SAS) Feature
The device's Stand-Alone Survivability (SAS) feature ensures telephony communication
continuity (survivability) for enterprises using hosted IP services (such as IP Centrex) or IPPBX in cases of failure of these entities. In case of failure of the IP Centrex, IP-PBX servers
(or even WAN connection and access Internet modem), the enterprise typically loses its
internal telephony service at any branch, between its offices, as well as with the external
environment. In addition, typically these failures lead to the inability to make emergency
calls (e.g., 911 in North America). Despite these possible point of failures, the device's SAS
feature ensures that the Enterprise's telephony services (e.g., SIP IP phones or soft
phones) are maintained by routing calls to the PSTN (i.e., providing PSTN fallback).
The SAS feature operates in one of two modes:
„
Normal: Initially, the device's SAS agent serves as a registrar (and outbound Proxy
server) to which every VoIP CPE (e.g., IP phones) within the Enterprise's LAN
registers. The SAS agent at the same time sends all these registration requests to the
Proxy server (e.g., IP-Centrex or IP-PBX). This ensures registration redundancy by the
SAS agent for all telephony devices. Therefore, SAS agent functions as a stateful
proxy, passing all SIP requests received from the Enterprise to the Proxy and vice
versa. In parallel, the SAS agent continuously maintains a keep-alive "handshake" with
the Proxy server using SIP OPTIONS or re-INVITE messages.
„
Emergency: The SAS agent switches to Emergency mode if it detects (from the keepalive responses) that the connection with the Proxy is lost. This can occur due to Proxy
server failure or WAN problems. In this mode, when the connection with the Proxy
server is down, the SAS agent controls all internal calls within the Enterprise. In the
case of outgoing calls, the SAS agent forwards them to a local VoIP gateway (this can
be the device itself or a separate analog or digital gateway). For PSTN fallback, the
local VoIP gateway should be equipped with analog (FXO) lines or digital (E1/T1)
trunk(s) for PSTN connectivity. In this way, the Enterprise preserves its capability for
internal and outgoing calls.
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The SAS agent continuously attempts to communicate with the Proxy using the regular
keep-alive method. After the connection is re-established, the SAS agent switches to
pre-Normal mode. In this mode, the SAS agent maintains all terminations of existing
calls while any new SIP call signaling (issued by new INVITE sessions) is transacted
to/from the Proxy server. This is accomplished using the SAS agent's database of
current active calls. After releasing all calls established during Emergency mode, the
SAS agent resumes operating in Normal mode.
For SAS implementation, the primary Proxy server for the VoIP CPE's (e.g., IP phones) is
the SAS agent (i.e., the device itself) while the IP Centrex or IP-PBX is defined as the
secondary Proxy server. For SAS configuration, the device is composed of two different
applications (SAS and Gateway), where each application has its own SIP interface
(UDP/TCP/TLS ports).
7.3.1
„
Configuring the device to use and operate with the SAS capabilities (refer to
''Configuring SAS'' on page 347)
„
Configuring SAS emergency call routing (refer to ''Configuring Emergency Calls'' on
page 348)
Configuring SAS
For configuring the device to operate with SAS, perform the following configurations:
„
IsProxyUsed = 1
„
ProxyIP 0 = <SAS agent's IP address, i.e., the device>
„
ProxyIP 1 = <external Proxy server IP address>
„
IsRegisterNeeded = 1 (for the device)
„
RegistrarIP = ‘ ‘
„
SIPDestinationPort = 5080
„
IsUserPhone = 0 (don’t use “user=phone” in SIP URL)
„
IsUserPhoneInFrom = 0 (don’t use “user=phone” in From Header)
„
IsFallbackUsed = 0
„
EnableProxyKeepAlive = 1 (enables keep-alive with Proxy using OPTIONS)
„
EnableSAS = 1
„
SASLocalSIPUDPPort = (default 5080)
„
SASRegistrationTime = <expiration time that SAS returns in the 200 OK to REGISTER
in Emergency mode> (default 20)
„
SASDefaultGatewayIP = < SAS gateway IP address>
„
SASProxySet = 1
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7.3.2
Configuring Emergency Calls
The device's SAS agent can be configured to detect a user-defined, emergency number
(e.g. 911 in North America), which it then redirects the call directly to the PSTN (through its
E1/T1 trunk). The emergency number is configured using the ini file parameter
SASEmergencyNumbers (for a detailed description, refer to ''SIP Configuration
Parameters'' on page 284).
Figure 7-1: Device's SAS Agent Redirecting Emergency Calls to PSTN
To configure support for emergency calls, configure the parameters below. The device and
the SAS feature are configured independently. If the device and the SAS agent use
different proxies, then the device's proxy server is defined using the 'Use Default Proxy'
parameter, while the SAS proxy agent is defined using the 'Proxy Set' table and
SASProxySet parameter.
„
EnableSAS = 1
„
SASLocalSIPUDPPort = (default 5080)
„
IsProxyUsed = 1
„
ProxyIP 0 = <external proxy IP address (device)>
„
ProxyIP 1 = <external proxy IP address (SAS)>
„
IsRegisterNeeded = 1 (for the device)
„
IsFallbackUsed = 0
„
SASRegistrationTime = <expiration time that SAS returns in the 200 OK to REGISTER
in Emergency mode> (default 20)
„
SASDefaultGatewayIP = < SAS gateway IP address>
„
SASProxySet = 1
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7. IP Telephony Capabilities
Configuring the DTMF Transport Types
You can control the way DTMF digits are transported over the IP network to the remote
endpoint, by using one of the following modes:
„
Using INFO message according to Nortel IETF draft: DTMF digits are carried to the
remote side in INFO messages. To enable this mode, define the following:
•
RxDTMFOption = 0 (ini file); 'Declare RFC 2833 in SDP' field = 'No' (Web
interface -- refer to ''DTMF & Dialing Parameters'' on page 147)
•
TxDTMFOption = 1 (ini file); '1st to 5th Tx DTMF Option' field = 'INFO (Nortel)' (Web
interface -- refer to ''DTMF & Dialing Parameters'' on page 147)
Note that in this mode, DTMF digits are erased from the audio stream
[DTMFTransportType is automatically set to 0 ('DTMF Transport Type' field = 'DTMF
Mute' -- Web interface)].
„
Using INFO message according to Cisco’s mode: DTMF digits are carried to the
remote side in INFO messages. To enable this mode, define the following:
•
RxDTMFOption = 0 (ini file); 'Declare RFC 2833 in SDP' field = 'No' (Web
interface -- refer to ''DTMF & Dialing Parameters'' on page 147)
•
TxDTMFOption = 3 (ini file); '1st to 5th Tx DTMF Option' field = 'INFO (Cisco)' (Web
interface -- refer to ''DTMF & Dialing Parameters'' on page 147)
Note that in this mode, DTMF digits are erased from the audio stream
(DTMFTransportType is automatically set to 0 ('DTMF Transport Type' field = 'DTMF
Mute' -- Web interface)].
„
Using NOTIFY messages according to <draft-mahy-sipping-signaled-digits01.txt>: DTMF digits are carried to the remote side using NOTIFY messages. To
enable this mode, define the following:
•
RxDTMFOption = 0 (ini file); 'Declare RFC 2833 in SDP' field = 'No' (Web
interface -- refer to ''DTMF & Dialing Parameters'' on page 147)
•
TxDTMFOption = 2 (ini file); '1st to 5th Tx DTMF Option' field = 'NOTIFY' (Web
interface -- refer to ''DTMF & Dialing Parameters'' on page 147)
Note that in this mode, DTMF digits are erased from the audio stream
(DTMFTransportType is automatically set to 0 ('DTMF Transport Type' field = 'DTMF
Mute' -- Web interface)].
„
Using RFC 2833 relay with Payload type negotiation: DTMF digits are carried to
the remote side as part of the RTP stream in accordance with RFC 2833 standard. To
enable this mode, define the following:
•
RxDTMFOption = 3 (ini file); 'Declare RFC 2833 in SDP' field = 'Yes' (Web
interface -- refer to ''DTMF & Dialing Parameters'' on page 147)
•
TxDTMFOption = 4 (ini file); '1st to 5th Tx DTMF Option' field = 'RFC 2833' (Web
interface -- refer to ''DTMF & Dialing Parameters'' on page 147)
Note that to set the RFC 2833 payload type with a different value (other than its
default, 96) configure the RFC2833PayloadType (RFC 2833 Payload Type)
parameter. The device negotiates the RFC 2833 payload type using local and remote
SDP and sends packets using the payload type from the received SDP. The device
expects to receive RFC 2833 packets with the same payload type as configured by the
RFC2833PayloadType parameter. If the remote side doesn’t include ‘telephony-event’
in its SDP, the device sends DTMF digits in transparent mode (as part of the voice
stream).
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„
„
Sending DTMF digits (in RTP packets) as part of the audio stream (DTMF Relay
is disabled): This method is typically used with G.711 coders; with other low-bit rate
(LBR) coders, the quality of the DTMF digits is reduced. To enable this mode, define
the following:
•
RxDTMFOption = 0 (ini file); 'Declare RFC 2833 in SDP' field = 'No' (Web
interface -- refer to ''DTMF & Dialing Parameters'' on page 147)
•
TxDTMFOption = 0 (ini file); '1st to 5th Tx DTMF Option' field = 'Disable' (Web
interface -- refer to ''DTMF & Dialing Parameters'' on page 147)
•
DTMFTransportType = 2 (DTMF Transport Type = Transparent DTMF)
Using INFO message according to Korea mode: DTMF digits are carried to the
remote side in INFO messages. To enable this mode, define the following:
•
RxDTMFOption = 0 (ini file); 'Declare RFC 2833 in SDP' field = 'No' (Web
interface -- refer to ''DTMF & Dialing Parameters'' on page 147)
•
TxDTMFOption = 3 (ini file); '1st to 5th Tx DTMF Option' field = 'INFO (Korea)' (Web
interface -- refer to ''DTMF & Dialing Parameters'' on page 147)
Note that in this mode, DTMF digits are erased from the audio stream
(DTMFTransportType is automatically set to 0 (DTMF Mute)).
Notes:
•
The device is always ready to receive DTMF packets over IP in all
possible transport modes: INFO messages, NOTIFY, and RFC 2833 (in
proper payload type) or as part of the audio stream.
•
To exclude RFC 2833 Telephony event parameter from the device's
SDP, set RxDTMFOption to 0 in the ini file.
The following parameters affect the way the device handles the DTMF digits:
„
TxDTMFOption, RxDTMFOption, and RFC2833PayloadType (described in ''DTMF &
Dialing Parameters'' on page 147)
„
MGCPDTMFDetectionPoint, DTMFVolume, DTMFTransportType, DTMFDigitLength,
and DTMFInterDigitInterval (refer to ''Channel Parameters'' on page 324)
7.5
Fax and Modem Capabilities
7.5.1
Fax/Modem Operating Modes
The device supports two modes of operations:
„
Fax / modem negotiation that isn’t performed during the establishment of the call.
„
VBD mode for V.152 implementation (refer to ''Supporting V.152 Implementation'' on
page 357): fax / modem capabilities are negotiated between the device and the remote
endpoint at the establishment of the call. During a call, when a fax / modem signal is
detected, transition from voice to VBD (or T.38) is automatically performed and no
additional SIP signaling is required. If negotiation fails (i.e., no match is achieved for
any of the transport capabilities), fallback to existing logic occurs (according to the
parameter IsFaxUsed).
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7.5.2
7. IP Telephony Capabilities
Fax/Modem Transport Modes
The device supports the following transport modes for fax per modem type
(V.22/V.23/Bell/V.32/V.34):
„
T.38 fax relay (refer to ''Fax Relay Mode'' on page 351)
„
Fax and modem bypass: a proprietary method that uses a high bit rate coder (refer to
''Fax/Modem Bypass Mode'' on page 352)
„
NSE Cisco’s Pass-through bypass mode for fax and modem (refer to ''Fax / Modem
NSE Mode'' on page 353)
„
Transparent: passing the fax / modem signal in the current voice coder (refer to ''Fax /
Modem Transparent Mode'' on page 354)
„
Transparent with events: passing the fax / modem signal in the current voice coder
with adaptations (refer to ''Fax / Modem Transparent with Events Mode'' on page 355)
„
G.711 Transport: switching to G.711 when fax/modem is detected (refer to ''G.711 Fax
/ Modem Transport Mode'' on page 355)
„
Fax fallback to G.711 if T.38 is not supported (refer to ''Fax Fallback'' on page 356)
‘Adaptations’ refer to automatic reconfiguration of certain DSP features for handling
fax/modem streams differently than voice.
7.5.2.1
T.38 Fax Relay Mode
In Fax Relay mode, fax signals are transferred using the T.38 protocol. T.38 is an ITU
standard for sending fax across IP networks in real-time mode. The device currently
supports only the T.38 UDP syntax.
T.38 can be configured in the following ways:
„
Switching to T.38 mode using SIP Re-INVITE messages (refer to ''Switching to T.38
Mode using SIP Re-INVITE'' on page 351)
„
Automatically switching to T.38 mode without using SIP Re-INVITE messages (refer to
''Automatically Switching to T.38 Mode without SIP Re-INVITE'' on page 352)
When fax transmission ends, the reverse switching from fax relay to voice is automatically
performed at both the local and remote endpoints.
You can change the fax rate declared in the SDP, using the parameter FaxRelayMaxRate
(this parameter doesn’t affect the actual transmission rate). In addition, you can enable or
disable Error Correction Mode (ECM) fax mode using the FaxRelayECMEnable parameter.
When using T.38 mode, you can define a redundancy feature to improve fax transmission
over congested IP networks. This feature is activated using the FaxRelayRedundancyDepth
and FaxRelayEnhancedRedundancyDepth parameters. Although this is a proprietary
redundancy scheme, it should not create problems when working with other T.38 decoders.
7.5.2.1.1 Switching to T.38 Mode using SIP Re-INVITE
In the Switching to T.38 Mode using SIP Re-INVITE mode, upon detection of a fax signal,
the terminating device negotiates T.38 capabilities using a Re-INVITE message. If the farend device doesn't support T.38, the fax fails. In this mode, the parameter
FaxTransportMode is ignored.
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To configure T.38 mode using SIP Re-INVITE messages, set IsFaxUsed to 1. Additional
configuration parameters include the following:
„
FaxRelayEnhancedRedundancyDepth
„
FaxRelayRedundancyDepth
„
FaxRelayECMEnable
„
FaxRelayMaxRate
7.5.2.1.2 Automatically Switching to T.38 Mode without SIP Re-INVITE
In the Automatically Switching to T.38 Mode without SIP Re-INVITE mode, when a fax
signal is detected, the channel automatically switches from the current voice coder to
answer tone mode, and then to T.38-compliant fax relay mode.
To configure automatic T.38 mode, perform the following configurations:
7.5.2.2
„
IsFaxUsed = 0
„
FaxTransportMode = 1
„
Additional configuration parameters:
•
FaxRelayEnhancedRedundancyDepth
•
FaxRelayRedundancyDepth
•
FaxRelayECMEnable
•
FaxRelayMaxRate
Fax/Modem Bypass Mode
In this proprietary mode, when fax or modem signals are detected, the channel
automatically switches from the current voice coder to a high bit-rate coder (according to
the parameter FaxModemBypassCoderType). In addition, the channel is automatically
reconfigured with the following fax / modem adaptations:
„
Disables silence suppression
„
Enables echo cancellation for fax
„
Disables echo cancellation for modem
„
Performs certain jitter buffering optimizations
The network packets generated and received during the bypass period are regular voice
RTP packets (per the selected bypass coder), but with a different RTP payload type
(according to the parameters FaxBypassPayloadType and ModemBypassPayloadType).
During the bypass period, the coder uses the packing factor, which is defined by the
parameter FaxModemBypassM. The packing factor determines the number of coder
payloads (each the size of FaxModemBypassBasicRTPPacketInterval) that are used to
generate a single fax/modem bypass packet. When fax/modem transmission ends, the
reverse switching, from bypass coder to regular voice coder is performed.
To configure fax / modem bypass mode, perform the following configurations:
„
IsFaxUsed = 0
„
FaxTransportMode = 2
„
V21ModemTransportType = 2
„
V22ModemTransportType = 2
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„
V23ModemTransportType = 2
„
V32ModemTransportType = 2
„
V34ModemTransportType = 2
„
BellModemTransportType = 2
„
Additional configuration parameters:
•
FaxModemBypassCoderType
•
FaxBypassPayloadType
•
ModemBypassPayloadType
•
FaxModemBypassBasicRTPPacketInterval
•
FaxModemBypassDJBufMinDelay
Note: When the device is configured for modem bypass and T.38 fax, V.21 lowspeed modems are not supported and fail as a result.
Tip:
7.5.2.3
When the remote (non-AudioCodes’) gateway uses G711 coder for voice and
doesn’t change the coder payload type for fax or modem transmission, it is
recommended to use the Bypass mode with the following configuration:
•
EnableFaxModemInbandNetworkDetection = 1
•
FaxModemBypassCoderType = same coder used for voice
•
FaxModemBypassM = same interval as voice
•
ModemBypassPayloadType = 8 if voice coder is A-Law; 0 if voice coder
is Mu-Law
Fax / Modem NSE Mode
In this mode, fax and modem signals are transferred using Cisco-compatible Pass-through
bypass mode. Upon detection of fax or modem answering tone signal, the terminating
device sends three to six special NSE RTP packets (using NSEpayloadType, usually 100).
These packets signal the remote device to switch to G.711 coder (according to the
parameter FaxModemBypassCoderType). After a few NSE packets are exchanged
between the devices, both devices start using G.711 packets with standard payload type (8
for G.711 A-Law and 0 for G.711 Mu-Law). In this mode, no Re-INVITE messages are sent.
The voice channel is optimized for fax/modem transmission (same as for usual bypass
mode).
The parameters defining payload type for the proprietary AudioCodes’ Bypass mode
FaxBypassPayloadType and ModemBypassPayloadType are not used with NSE Bypass.
When configured for NSE mode, the device includes in its SDP the following line:
a=rtpmap:100 X-NSE/8000
(where 100 is the NSE payload type)
The Cisco gateway must include the following definition: "modem passthrough nse payloadtype 100 codec g711alaw".
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To configure NSE mode, perform the following configurations:
7.5.2.4
„
IsFaxUsed = 0
„
FaxTransportMode = 2
„
NSEMode = 1
„
NSEPayloadType = 100
„
V21ModemTransportType = 2
„
V22ModemTransportType = 2
„
V23ModemTransportType = 2
„
V32ModemTransportType = 2
„
V34ModemTransportType = 2
„
BellModemTransportType = 2
Fax / Modem Transparent Mode
In this mode, fax and modem signals are transferred using the current voice coder without
notifications to the user and without automatic adaptations. It's possible to use the Profiles
mechanism (refer to ''Configuring the Profile Definitions'' on page 190) to apply certain
adaptations to the channel used for fax / modem (e.g., to use the coder G.711, to set the
jitter buffer optimization factor to 13, and to enable echo cancellation for fax and disable it
for modem).
To configure fax / modem transparent mode, use the following parameters:
„
IsFaxUsed = 0
„
FaxTransportMode = 0
„
V21ModemTransportType = 0
„
V22ModemTransportType = 0
„
V23ModemTransportType = 0
„
V32ModemTransportType = 0
„
V34ModemTransportType = 0
„
BellModemTransportType = 0
„
Additional configuration parameters:
•
CoderName
•
DJBufOptFactor
•
EnableSilenceCompression
•
EnableEchoCanceller
Note: This mode can be used for fax, but is not recommended for modem
transmission. Instead, use the modes Bypass (refer to ''Fax/Modem Bypass
Mode'' on page 352) or Transparent with Events (refer to ''Fax / Modem
Transparent with Events Mode'' on page 355) for modem.
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Fax / Modem Transparent with Events Mode
In this mode, fax and modem signals are transferred using the current voice coder with the
following automatic adaptations:
„
Echo Canceller = on (or off, for modems)
„
Echo Canceller Non-Linear Processor Mode = off
„
Jitter buffering optimizations
To configure fax / modem transparent with events mode, perform the following
configurations:
7.5.2.6
„
IsFaxUsed = 0
„
FaxTransportMode = 3
„
V21ModemTransportType = 3
„
V22ModemTransportType = 3
„
V23ModemTransportType = 3
„
V32ModemTransportType = 3
„
V34ModemTransportType = 3
„
BellModemTransportType = 3
G.711 Fax / Modem Transport Mode
In this mode, when the terminating device detects fax or modem signals (CED or AnsAM), it
sends a Re-INVITE message to the originating device requesting it to re-open the channel
in G.711 VBD with the following adaptations:
„
Echo Canceller = off
„
Silence Compression = off
„
Echo Canceller Non-Linear Processor Mode = off
„
Dynamic Jitter Buffer Minimum Delay = 40
„
Dynamic Jitter Buffer Optimization Factor = 13
After a few seconds upon detection of fax V.21 preamble or super G3 fax signals, the
device sends a second Re-INVITE enabling the echo canceller (the echo canceller is
disabled only on modem transmission).
A ‘gpmd’ attribute is added to the SDP according to the following format:
„
For G.711A-law: a=gpmd:0 vbd=yes;ecan=on (or off, for modems)
„
For G.711 µ-law: a=gpmd:8 vbd=yes;ecan=on (or off for modems)
The parameters FaxTransportMode and VxxModemTransportType are ignored and
automatically set to the mode called ‘transparent with events’.
To configure fax / modem transparent mode, set IsFaxUsed to 2.
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7.5.2.7
Fax Fallback
In this mode, when the terminating device detects a fax signal, it sends a Re-INVITE
message to the originating device with T.38. If the remote device doesn’t support T.38
(replies with SIP response 415 'Media Not Supported'), the device sends a new Re-INVITE
with G.711 VBD with the following adaptations:
„
Echo Canceller = on
„
Silence Compression = off
„
Echo Canceller Non-Linear Processor Mode = off
„
Dynamic Jitter Buffer Minimum Delay = 40
„
Dynamic Jitter Buffer Optimization Factor = 13
When the device initiates a fax session using G.711, a ‘gpmd’ attribute is added to the SDP
according to the following format:
„
For G.711A-law: a=gpmd:0 vbd=yes;ecan=on
„
For G.711 µ-law: a=gpmd:8 vbd=yes;ecan=on
In this mode, the parameter FaxTransportMode is ignored and automatically set to
‘transparent’.
To configure fax fallback mode, set IsFaxUsed to 3.
7.5.3
Supporting V.34 Faxes
Unlike T.30 fax machines, V.34 fax machines have no relay standard to transmit data over
IP to the remote side. Therefore, the device provides the following operation modes for
transporting V.34 fax data over the IP:
„
Using bypass mechanism for V.34 fax transmission (refer to ''Using Bypass
Mechanism for V.34 Fax Transmission'' on page 356)
„
Using relay mode, i.e., fallback to T.38 (refer to ''Using Relay mode for both T.30 and
V.34 faxes'' on page 357)
Using the ini file parameter V34FaxTransportType, you can determine whether to pass V.34
Fax-over-T.38 fallback to T.30, or use Bypass over the High Bit Rate coder (e.g. PCM ALaw).
Note: The CNG detector is disabled (CNGDetectorMode = 0) in all the subsequent
examples.
7.5.3.1
Using Bypass Mechanism for V.34 Fax Transmission
In this proprietary scenario, the device uses bypass (or NSE) mode to transmit V.34 faxes,
enabling the full utilization of its speed.
Configure the following parameters to use bypass mode for both T.30 and V.34 faxes:
„
FaxTransportMode = 2 (Bypass)
„
V34ModemTransportType = 2 (Modem bypass)
„
V32ModemTransportType = 2
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„
V23ModemTransportType = 2
„
V22ModemTransportType = 2
7. IP Telephony Capabilities
Configure the following parameters to use bypass mode for V.34 faxes and T.38 for T.30
faxes:
7.5.3.2
„
FaxTransportMode = 1 (Relay)
„
V34ModemTransportType = 2 (Modem bypass)
„
V32ModemTransportType = 2
„
V23ModemTransportType = 2
„
V22ModemTransportType = 2
Using Relay mode for both T.30 and V.34 faxes
In this scenario, V.34 fax machines are forced to use their backward compatibility with T.30
faxes and operate in the slower T.30 mode.
Use the following parameters to use T.38 mode for both V.34 faxes and T.30 faxes:
7.5.4
„
FaxTransportMode = 1 (Relay)
„
V34ModemTransportType = 0 (Transparent)
„
V32ModemTransportType = 0
„
V23ModemTransportType = 0
„
V22ModemTransportType = 0
Supporting V.152 Implementation
The device supports the ITU-T recommendation V.152 (Procedures for Supporting VoiceBand Data over IP Networks). Voice-band data (VBD) is the transport of modem, facsimile,
and text telephony signals over a voice channel of a packet network with a codec
appropriate for such signals.
For V.152 capability, the device supports T.38 as well as VBD codecs (i.e., G.711 A-law
and G.711 μ-law). The selection of capabilities is performed using the coders table (refer to
''Coders'' on page 144).
When in VBD mode for V.152 implementation, support is negotiated between the device
and the remote endpoint at the establishment of the call. During this time, initial exchange
of call capabilities is exchanged in the outgoing SDP. These capabilities include whether
VBD is supported and associated RTP payload types ('gpmd' SDP attribute), supported
codecs, and packetization periods for all codec payload types ('ptime' SDP attribute). After
this initial negotiation, no Re-INVITE messages are necessary as both endpoints are
synchronized in terms of the other side's capabilities. If negotiation fails (i.e., no match was
achieved for any of the transport capabilities), fallback to existing logic occurs (according to
the parameter IsFaxUsed).
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Below is an example of media descriptions of an SDP indicating support for V.152.
v=0
o=- 0 0 IN IPV4 <IPAdressA>
s=t=0 0
p=+1
c=IN IP4 <IPAddressA
m=audio <udpPort A> RTP/AVP 18 0
a=ptime:10
a=rtpmap:96 PCMU/8000
a=gpmd: 96 vbd=yes
In the example above, V.152 implementation is supported (using the dynamic payload type
96 and G.711 u-law as the VBD codec) as well as the voice codecs G.711 μ-law and
G.729.
Instead of using VBD transport mode, the V.152 implementation can use alternative relay
fax transport methods (e.g., fax relay over IP using T.38). The preferred V.152 transport
method is indicated by the SDP ‘pmft’ attribute. Omission of this attribute in the SDP
content means that VBD mode is the preferred transport mechanism for voice-band data.
To configure T.38 mode, use the CoderName parameter.
7.6
Event Notification using X-Detect Header
The device supports the sending of notifications to a remote party notifying the occurrence
(or detection) of certain events on the media stream. Event detection and notifications is
performed using the X-Detect SIP message header, and only when establishing a SIP
dialog.
For supporting some events, certain device configurations need to be performed. The table
below lists the support event types (and subtypes) and the corresponding device
configurations, if required:
Table 7-2: Supported X-Detect Event Types
Events Type
Subtype
AMD
voice
automatic
silence
unknown
Required Configuration
EnableDSPIPMDetectors = 1
AMDTimeout = 2000 (msec)
CPT
SIT
SITDetectorEnable = 1
UserDefinedToneDetectorEnable = 1
FAX
CED
(IsFaxUsed ≠ 0) or (IsFaxUsed = 0, and
FaxTransportMode ≠ 0)
PTT
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VxxModemTransportType = 3
voice-start
voice-end
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The X-Detect event notification process is as follows:
1.
For IP-to-Tel or Tel-to-IP calls, the device receives a SIP request message (using the
X-Detect header) that the remote party wishes to detect events on the media stream.
For incoming (IP-to-Tel) calls, the request must be indicated in the initial INVITE and
responded to either in the 183 response (for early dialogs) or in the 200 OK response
(for confirmed dialogs). For outgoing calls (Tel-to-IP), the request may be received in
the 183 (for early dialogs) and responded to in the PRACK, or received in the 200 OK
(for confirmed dialogs) and responded to in the ACK.
2.
Once the device receives such a request, it sends a SIP response message (using the
X-Detect header) to the remote party, listing all supported events that can be detected.
The absence of the X-Detect header indicates that no detections are available.
3.
Each time the device detects a supported event, the event is notified to the remote
party, by sending an INFO message with the following message body:
•
Content-Type: application/X-DETECT
•
Type = [AMD | CPT | FAX | PTT…]
•
Subtype = xxx (according to the defined subtypes of each type)
Below is an example of SIP messages implementing the X-Detect header:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.33.2.53;branch=z9hG4bKac5906
Max-Forwards: 70
From: "anonymous" <sip:[email protected]>;tag=1c25298
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
X- Detect: Request=CPT,FAX
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.33.2.53;branch=z9hG4bKac5906
From: "anonymous" <sip:[email protected]>;tag=1c25298
To: <sip:[email protected];user=phone>;tag=1c19282
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
X- Detect: Response=CPT,FAX
INFO sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.33.2.53;branch=z9hG4bKac5906
Max-Forwards: 70
From: "anonymous" <sip:[email protected]>;tag=1c25298
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
X- Detect: Response=CPT,FAX
Content-Type: Application/X-Detect
Content-Length: xxx
Type = CPT
Subtype = SIT
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7.7
RTP Multiplexing (ThroughPacket)
The device supports a proprietary method to aggregate RTP streams from several channels
to reduce the bandwidth overhead caused by the attached Ethernet, IP, UDP, and RTP
headers, and to reduce the packet / data transmission rate. This option reduces the load on
network routers and can typically save 50% (e.g., for G.723) on IP bandwidth. RTP
Multiplexing (ThroughPacket™) is accomplished by aggregating payloads from several
channels that are sent to the same destination IP address into a single IP packet.
RTP multiplexing can be applied to the entire device (refer to ''Configuring the RTP / RTCP
Settings'' on page 71) or to specific IP destinations using the IP Profile feature (refer to ''IP
Profile Settings'' on page 193).
To enable RTP Multiplexing, set the parameter RemoteBaseUDPPort to a nonzero value.
Note that the value of RemoteBaseUDPPort on the local device must equal the value of
BaseUDPPort of the remote device. The device uses these parameters to identify and
distribute the payloads from the received multiplexed IP packet to the relevant channels.
In RTP Multiplexing mode, the device uses a single UDP port for all incoming multiplexed
packets and a different port for outgoing packets. These ports are configured using the
parameters L1L1ComplexTxUDPPort and L1L1ComplexRxUDPPort.
When RTP Multiplexing is used, call statistics aren’t available (since there is no RTCP
flow).
Notes:
7.8
•
RTP Multiplexing must be enabled on both devices.
•
When VLANs are imlemented, the RTP Multiplexing mechanism is not
supported.
Dynamic Jitter Buffer Operation
Voice frames are transmitted at a fixed rate. If the frames arrive at the other end at the
same rate, voice quality is perceived as good. In many cases, however, some frames can
arrive slightly faster or slower than the other frames. This is called jitter (delay variation),
and degrades the perceived voice quality. To minimize this problem, the device uses a jitter
buffer. The jitter buffer collects voice packets, stores them and sends them to the voice
processor in evenly spaced intervals.
The device uses a dynamic jitter buffer that can be configured using the following two
parameters:
„
Minimum delay: DJBufMinDelay (0 msec to 150 msec)
Defines the starting jitter capacity of the buffer. For example, at 0 msec, there is no
buffering at the start. At the default level of 10 msec, the device always buffers
incoming packets by at least 10 msec worth of voice frames.
„
Optimization Factor: DJBufOptFactor (0 to 12, 13)
Defines how the jitter buffer tracks to changing network conditions. When set at its
maximum value of 12, the dynamic buffer aggressively tracks changes in delay (based
on packet loss statistics) to increase the size of the buffer and doesn’t decay back
down. This results in the best packet error performance, but at the cost of extra delay.
At the minimum value of 0, the buffer tracks delays only to compensate for clock drift
and quickly decays back to the minimum level. This optimizes the delay performance
but at the expense of a higher error rate.
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The default settings of 10 msec Minimum delay and 10 Optimization Factor should provide
a good compromise between delay and error rate. The jitter buffer ‘holds’ incoming packets
for 10 msec before making them available for decoding into voice. The coder polls frames
from the buffer at regular intervals in order to produce continuous speech. As long as
delays in the network do not change (jitter) by more than 10 msec from one packet to the
next, there is always a sample in the buffer for the coder to use. If there is more than 10
msec of delay at any time during the call, the packet arrives too late. The coder tries to
access a frame and is not able to find one. The coder must produce a voice sample even if
a frame is not available. It therefore compensates for the missing packet by adding a BadFrame-Interpolation (BFI) packet. This loss is then flagged as the buffer being too small.
The dynamic algorithm then causes the size of the buffer to increase for the next voice
session. The size of the buffer may decrease again if the device notices that the buffer is
not filling up as much as expected. At no time does the buffer decrease to less than the
minimum size configured by the Minimum delay parameter.
For certain scenarios, the Optimization Factor is set to 13: One of the purposes of the
Jitter Buffer mechanism is to compensate for clock drift. If the two sides of the VoIP call are
not synchronized to the same clock source, one RTP source generates packets at a lower
rate, causing under-runs at the remote Jitter Buffer. In normal operation (optimization factor
0 to 12), the Jitter Buffer mechanism detects and compensates for the clock drift by
occasionally dropping a voice packet or by adding a BFI packet.
Fax and modem devices are sensitive to small packet losses or to added BFI packets.
Therefore, to achieve better performance during modem and fax calls, the Optimization
Factor should be set to 13. In this special mode the clock drift correction is performed less
frequently - only when the Jitter Buffer is completely empty or completely full. When such
condition occurs, the correction is performed by dropping several voice packets
simultaneously or by adding several BFI packets simultaneously, so that the Jitter Buffer
returns to its normal condition.
7.9
Configuring Alternative Routing (Based on
Connectivity and QoS)
The Alternative Routing feature enables reliable routing of Tel-to-IP calls when a Proxy isn’t
used. The device periodically checks the availability of connectivity and suitable Quality of
Service (QoS) before routing. If the expected quality cannot be achieved, an alternative IP
route for the prefix (phone number) is selected.
Note: If the alternative routing destination is the device itself, the call can be
configured to be routed back to one of the device's trunk groups and thus,
back into the PSTN (PSTN Fallback).
7.9.1
Alternative Routing Mechanism
When a Tel-to-IP call is routed through the device, the call’s destination number is
compared to the list of prefixes defined in the 'Tel to IP Routing' table (described in ''Tel to
IP Routing Table'' on page 175). The 'Tel to IP Routing' table is scanned for the destination
number’s prefix starting at the top of the table. For this reason, enter the main IP route
above any alternative route. When an appropriate entry (destination number matches one
of the prefixes) is found, the prefix’s corresponding destination IP address is verified. If the
destination IP address is disallowed (or if the original call fails and the device has made two
additional attempts to establish the call without success), an alternative route is searched in
the table. , after which an alternative route is used.
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Destination IP address is disallowed if no ping to the destination is available (ping is
continuously initiated every seven seconds), when an inappropriate level of QoS was
detected, or when a DNS host name is not resolved. The QoS level is calculated according
to delay or packet loss of previously ended calls. If no call statistics are received for two
minutes, the QoS information is reset.
7.9.2
Determining the Availability of Destination IP Addresses
To determine the availability of each destination IP address (or host name) in the routing
table, one (or all) of the following (configurable) methods are applied:
7.9.3
„
Connectivity: The destination IP address is queried periodically (currently only by
ping).
„
QoS: The QoS of an IP connection is determined according to RTCP statistics of
previous calls. Network delay (in msec) and network packet loss (in percentage) are
separately quantified and compared to a certain (configurable) threshold. If the
calculated amounts (of delay or packet loss) exceed these thresholds, the IP
connection is disallowed.
„
DNS resolution: When host name is used (instead of IP address) for the destination
route, it is resolved to an IP address by a DNS server. Connectivity and QoS are then
applied to the resolved IP address.
PSTN Fallback as a Special Case of Alternative Routing
The PSTN Fallback feature enables the device to redirect PSTN originated calls back to the
legacy PSTN network if a destination IP route is unsuitable (disallowed) for voice traffic at a
specific time. To enable PSTN fallback, assign the device's IP address as an alternative
route to the desired prefixes. Note that calls (now referred to as IP-to-Tel calls) can be rerouted to a specific trunk group using the Routing parameters (refer to ''IP to Trunk Group
Routing'' on page 181).
7.9.4
Relevant Parameters
The following parameters (described in ''Routing General Parameters'' on page 171) are
used to configure the Alternative Routing mechanism:
„
AltRoutingTel2IPEnable
„
AltRoutingTel2IPMode
„
IPConnQoSMaxAllowedPL
„
IPConnQoSMaxAllowedDelay
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7. IP Telephony Capabilities
Supported RADIUS Attributes
Use the following table for explanations on the RADIUS attributes contained in the
communication packets transmitted between the device and a RADIUS Server.
Table 7-3: Supported RADIUS Attributes
Attribute
Number
Attribute
Name
VSA
No.
Purpose
Value
Format
String
up to 15
digits
long
Example
AAA1
Request Attributes
1
User-Name
Account number or calling
party number or blank
4
NAS-IPAddress
IP address of the
requesting device
6
ServiceType
26
H323IncomingConf-Id
1
SIP call identifier
26
H323RemoteAddress
23
26
H323-ConfID
26
H323-SetupTime
26
H323-CallOrigin
Type of service requested
Numeric
Numeric
5421385747
Start
Acc
Stop
Acc
192.168.14.43
Start
Acc
Stop
Acc
1: login
Start
Acc
Stop
Acc
Up to
32
octets
Start
Acc
Stop
Acc
IP address of the remote
gateway
Numeric
Stop
Acc
24
H.323/SIP call identifier
Up to
32
octets
Start
Acc
Stop
Acc
25
Setup time in NTP format
1
String
Start
Acc
Stop
Acc
26
The call’s originator:
Answering (IP) or
Originator (PSTN)
26
H323-CallType
27
Protocol type or family
used on this leg of the call
String
26
H323ConnectTime
28
Connect time in NTP
format
String
Stop
Acc
26
H323-
29
Disconnect time in NTP
String
Stop
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Stop
Acc
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Attribute
Number
Attribute
Name
VSA
No.
DisconnectTime
26
26
H323DisconnectCause
H323-Gw-ID
26
SIP-Call-ID
26
CallTerminator
30
CalledStation-ID
Purpose
Value
Format
Example
format
30
33
AAA1
Acc
Q.931 disconnect cause
code
Name of the gateway
Numeric
Stop
Acc
String
SIPIDString
Start
Acc
Stop
Acc
34
SIP Call ID
String
[email protected]
Start
Acc
Stop
Acc
35
The call's terminator:
PSTN-terminated call
(Yes); IP-terminated call
(No).
String
Yes, No
Stop
Acc
String
8004567145
Start
Acc
String
2427456425
Stop
Acc
5135672127
Start
Acc
Stop
Acc
1: start, 2: stop
Start
Acc
Stop
Acc
5
Start
Acc
Stop
Acc
Destination phone
number
CallingStation-ID
Calling Party Number
(ANI)
40
Acct-StatusType
Account Request Type
(start or stop)
Note: ‘start’ isn’t
supported on the Calling
Card application.
41
Acct-DelayTime
No. of seconds tried in
sending a particular
record
Numeric
42
Acct-InputOctets
Number of octets
received for that call
duration
Numeric
Stop
Acc
43
Acct-OutputOctets
Number of octets sent for
that call duration
Numeric
Stop
Acc
44
AcctSession-ID
A unique accounting
identifier - match start &
stop
String
Start
Acc
Stop
Acc
46
AcctSessionTime
For how many seconds
the user received the
service
Numeric
Stop
Acc
47
Acct-InputPackets
Number of packets
received during the call
Numeric
Stop
Acc
31
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String
Numeric
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Attribute
Number
Attribute
Name
VSA
No.
48
Acct-OutputPackets
Number of packets sent
during the call
61
NAS-PortType
Physical port type of
device on which the call is
active
Purpose
Value
Format
Example
Stop
Acc
Numeric
String
AAA1
0:
Asynchronous
Start
Acc
Stop
Acc
0 Request
accepted
Stop
Acc
Response Attributes
26
H323ReturnCode
44
AcctSession-ID
103
The reason for failing
authentication (0 = ok,
other number failed)
A unique accounting
identifier – match start &
stop
Numeric
String
Stop
Acc
Below is an example of RADIUS Accounting, where the non-standard parameters are
preceded with brackets.
Accounting-Request (361)
user-name = 111
acct-session-id = 1
nas-ip-address = 212.179.22.213
nas-port-type = 0
acct-status-type = 2
acct-input-octets = 4841
acct-output-octets = 8800
acct-session-time = 1
acct-input-packets = 122
acct-output-packets = 220
called-station-id = 201
calling-station-id = 202
// Accounting non-standard parameters:
(4923 33) h323-gw-id =
(4923 23) h323-remote-address = 212.179.22.214
(4923 1) h323-ivr-out = h323-incoming-conf-id:02102944 600a1899
3fd61009 0e2f3cc5
(4923 30) h323-disconnect-cause = 22 (0x16)
(4923 27) h323-call-type = VOIP
(4923 26) h323-call-origin = Originate
(4923 24) h323-conf-id = 02102944 600a1899 3fd61009 0e2f3cc5
7.11
Call Detail Record
The Call Detail Record (CDR) contains vital statistic information on calls made by the
device. CDRs are generated at the end and (optionally) at the beginning of each call
(determined by the parameter CDRReportLevel), and then sent to a Syslog server. The
destination IP address for CDR logs is determined by the parameter CDRSyslogServerIP.
For CDR in RADIUS format, refer to ''Supported RADIUS Attributes'' on page 362.
The following table lists the CDR fields that are supported.
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Table 7-4: Supported CDR Fields
Field Name
Description
ReportType
Report for either Call Started, Call Connected, or Call Released
Cid
Port Number
CallId
SIP Call Identifier
Trunk
Physical Trunk Number
BChan
Selected B-Channel
ConId
SIP Conference ID
TG
Trunk Group Number
EPTyp
Endpoint Type
Orig
Call Originator (IP, Tel)
SourceIp
Source IP Address
DestIp
Destination IP Address
TON
Source Phone Number Type
NPI
Source Phone Number Plan
SrcPhoneNum
Source Phone Number
SrcNumBeforeMap
Source Number Before Manipulation
TON
Destination Phone Number Type
NPI
Destination Phone Number Plan
DstPhoneNum
Destination Phone Number
DstNumBeforeMap
Destination Number Before Manipulation
Durat
Call Duration
Coder
Selected Coder
Intrv
Packet Interval
RtpIp
RTP IP Address
Port
Remote RTP Port
TrmSd
Initiator of Call Release (IP, Tel, Unknown)
TrmReason
Termination Reason
Fax
Fax Transaction during the Call
InPackets
Number of Incoming Packets
OutPackets
Number of Outgoing Packets
PackLoss
Local Packet Loss
RemotePackLoss
Number of Outgoing Lost Packets
UniqueId
unique RTP ID
SetupTime
Call Setup Time
ConnectTime
Call Connect Time
ReleaseTime
Call Release Time
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Field Name
Description
RTPdelay
RTP Delay
RTPjitter
RTP Jitter
RTPssrc
Local RTP SSRC
RemoteRTPssrc
Remote RTP SSRC
RedirectReason
Redirect Reason
TON
Redirection Phone Number Type
MeteringPulses
Number of Generated Metering Pulses
NPI
Redirection Phone Number Plan
RedirectPhonNum
Redirection Phone Number
7.12
Trunk-to-Trunk Routing Example
This example describes two devices, each interfacing with the PSTN through four E1
spans. Device A is configured to route all incoming Tel-to-IP calls to Device B. Device B
generates calls to the PSTN on the same E1 trunk on which the call was originally received
(in Device A).
„
Device A IP address: 192.168.3.50
„
Device B IP address: 192.168.3.51
The ini file parameters configuration for devices A and B include the following:
1.
2.
At both devices, define four trunk groups, each with 30 B-channels:
•
TrunkGroup_1 = 0/1-31,1000
•
TrunkGroup_2 = 1/1-31,2000
•
TrunkGroup_3 = 2/1-31,3000
•
TrunkGroup_4 = 3/1-31,4000
At Device A, add the originating Trunk Group ID as a prefix to the destination number
for Tel-to-IP calls:
AddTrunkGroupAsPrefix = 1
3.
At Device A, route all incoming PSTN calls starting with prefixes 1, 2, 3, and 4, to the
IP address of Device B:
•
Prefix = 1, 192.168.3.51
•
Prefix = 2, 192.168.3.51
•
Prefix = 3, 192.168.3.51
•
Prefix = 4, 192.168.3.51
Note: You can also define Prefix = *,192.168.3.51, instead of the four lines above.
4.
Version 5.6
At Device B, route IP-to-PSTN calls to Trunk Group ID according to the first digit of the
called number:
•
PSTNPrefix = 1,1
•
PSTNPrefix = 2,2
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7.13
•
PSTNPrefix = 3,4
•
PSTNPrefix = 4,4
At Device B, remove the first digit from each IP-to-PSTN number before it is used in an
outgoing call: NumberMapIP2Tel = *,1
Proxy or Registrar Registration Example
Below is an example of Proxy and Registrar Registration:
REGISTER sip:servername SIP/2.0
VIA: SIP/2.0/UDP 212.179.22.229;branch=z9hG4bRaC7AU234
From: <sip:GWRegistrationName@sipgatewayname>;tag=1c29347
To: <sip:GWRegistrationName@sipgatewayname>
Call-ID: [email protected]
Seq: 1 REGISTER
Expires: 3600
Contact: sip:[email protected]
Content-Length: 0
The ‘servername’ string is defined according to the following rules:
„
The "servername" is equal to "RegistrarName" if configured. The "RegistrarName" can
be any string.
„
Otherwise, the "servername" is equal to "RegistrarIP" (either FQDN or numerical IP
address), if configured.
„
Otherwise, the "servername" is equal to "ProxyName" if configured. The "ProxyName"
can be any string.
„
Otherwise, the "servername" is equal to "ProxyIP" (either FQDN or numerical IP
address).
The parameter GWRegistrationName can be any string. This parameter is used only if
registration is per device. If the parameter is not defined, the parameter UserName is used
instead. If the registration is per endpoint, the endpoint phone number is used.
The 'sipgatewayname' parameter (defined in the ini file or Web interface) can be any string.
Some Proxy servers require that the 'sipgatewayname' (in REGISTER messages) is set
equal to the Registrar / Proxy IP address or to the Registrar / Proxy domain name. The
'sipgatewayname' parameter can be overwritten by the TrunkGroupSettings_GatewayName
value if the TrunkGroupSettings_RegistrationMode is set to 'Per Endpoint'.
REGISTER messages are sent to the Registrar's IP address (if configured) or to the Proxy's
IP address. A single message is sent once per device, or messages are sent per B-channel
according to the parameter AuthenticationMode. There is also an option to configure
registration mode per Trunk Group using the TrunkGroupSettings table. The registration
request is resent according to the parameter RegistrationTimeDivider. For example, if
RegistrationTimeDivider = 70 (%) and Registration Expires time = 3600, the device resends
its registration request after 3600 x 70% = 2520 sec. The default value of
RegistrationTimeDivider is 50%.
If registration per B-channel is selected, on device startup the device sends REGISTER
requests according to the maximum number of allowed SIP dialogs (configured by the
parameter NumberOfActiveDialogs). After each received response, the subsequent
REGISTER request is sent.
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7.14
7. IP Telephony Capabilities
Configuration Examples
7.14.1 SIP Call Flow
The SIP call flow (shown in the following figure), describes SIP messages exchanged
between two devices during a simple call. In this call flow example, device (10.8.201.158)
with phone number ‘6000’ dials device (10.8.201.161) with phone number ‘2000’.
Figure 7-2: SIP Call Flow
„ F1 (10.8.201.108 >> 10.8.201.10 INVITE):
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd
From: <sip:[email protected]>;tag=1c5354
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 18153 INVITE
Contact: <sip:[email protected];user=phone>
User-Agent: Audiocodes-Sip-Gateway/Mediant 2000/v.5.40.010.006
Supported: 100rel,em
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,
NOTIFY,PRACK,REFER,INFO
Content-Type: application/sdp
Content-Length: 208
v=0
o=AudiocodesGW 18132 74003 IN IP4 10.8.201.108
s=Phone-Call
c=IN IP4 10.8.201.108
t=0 0
m=audio 4000 RTP/AVP 8 96
a=rtpmap:8 pcma/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
„
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F2 (10.8.201.10 >> 10.8.201.108 TRYING):
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SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd
From: <sip:[email protected]>;tag=1c5354
To: <sip:[email protected]>
Call-ID: [email protected]
Server: Audiocodes-Sip-Gateway/Mediant 2000/v.5.40.010.006
CSeq: 18153 INVITE
Content-Length: 0
„
F3 (10.8.201.10 >> 10.8.201.108 180 RINGING):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd
From: <sip:[email protected]>;tag=1c5354
To: <sip:[email protected]>;tag=1c7345
Call-ID: [email protected]
Server: Audiocodes-Sip-Gateway/Mediant 2000/v.5.40.010.006
CSeq: 18153 INVITE
Supported: 100rel,em
Content-Length: 0
Note: Phone ‘1000’ answers the call and then sends a 200 OK message to device
10.8.201.108.
„
F4 (10.8.201.10 >> 10.8.201.108 200 OK):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd
From: <sip:[email protected]>;tag=1c5354
To: <sip:[email protected]>;tag=1c7345
Call-ID: [email protected]
CSeq: 18153 INVITE
Contact: <sip:[email protected];user=phone>
Server: Audiocodes-Sip-Gateway/Mediant 2000/v.5.40.010.006
Supported: 100rel,em
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,
NOTIFY,PRACK,REFER,INFO
Content-Type: application/sdp
Content-Length: 206
v=0
o=AudiocodesGW 30221 87035 IN IP4 10.8.201.10
s=Phone-Call
c=IN IP4 10.8.201.10
t=0 0
m=audio 7210 RTP/AVP 8 96
a=rtpmap:8 pcma/8000
a=ptime:20
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
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„
7. IP Telephony Capabilities
F5 (10.8.201.108 >> 10.8.201.10 ACK):
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacZYpJWxZ
From: <sip:[email protected]>;tag=1c5354
To: <sip:[email protected]>;tag=1c7345
Call-ID: [email protected]
User-Agent: Audiocodes-Sip-Gateway/Mediant 2000/v.5.40.010.006
CSeq: 18153 ACK
Supported: 100rel,em
Content-Length: 0
Note: Phone ‘8000’ goes on-hook and device 10.8.201.108 sends a BYE to device
10.8.201.10. Voice path is established.
„
F6 (10.8.201.108 >> 10.8.201.10 BYE):
BYE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacRKCVBud
From: <sip:[email protected]>;tag=1c5354
To: <sip:[email protected]>;tag=1c7345
Call-ID: [email protected]
User-Agent: Audiocodes-Sip-Gateway/Mediant 2000/v.5.40.010.006
CSeq: 18154 BYE
Supported: 100rel,em
Content-Length: 0
„
F7 (10.8.201.10 >> 10.8.201.108 200 OK):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacRKCVBud
From: <sip:[email protected]>;tag=1c5354
To: <sip:[email protected]>;tag=1c7345
Call-ID: [email protected]
Server: Audiocodes-Sip-Gateway/Mediant 2000/v.5.40.010.006
CSeq: 18154 BYE
Supported: 100rel,em
Content-Length: 0
7.14.2 SIP Authentication Example
The device supports basic and digest (MD5) authentication types, according to SIP RFC
3261 standard. A proxy server might require authentication before forwarding an INVITE
message. A Registrar/Proxy server may also require authentication for client registration. A
proxy replies to an unauthenticated INVITE with a 407 Proxy Authorization Required
response, containing a Proxy-Authenticate header with the form of the challenge. After
sending an ACK for the 407, the user agent can then resend the INVITE with a ProxyAuthorization header containing the credentials.
User agent, redirect or registrar servers typically use 401 Unauthorized response to
challenge authentication containing a WWW-Authenticate header, and expect the reINVITE to contain an Authorization header.
The following example describes the Digest Authentication procedure, including
computation of user agent credentials:
1.
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The REGISTER request is sent to Registrar/Proxy server for registration, as follows:
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REGISTER sip:10.2.2.222 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.200
From: <sip: [email protected]>;tag=1c17940
To: <sip: [email protected]>
Call-ID: [email protected]
User-Agent: Audiocodes-Sip-Gateway/Mediant 2000/v.5.40.010.006
CSeq: 1 REGISTER
Contact: sip:[email protected]:
Expires:3600
2.
Upon receipt of this request, the Registrar/Proxy returns 401 Unauthorized response.
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.2.1.200
From: <sip:[email protected] >;tag=1c17940
To: <sip:[email protected] >
Call-ID: [email protected]
Cseq: 1 REGISTER
Date: Mon, 30 Jul 2001 15:33:54 GMT
Server: Columbia-SIP-Server/1.17
Content-Length: 0
WWW-Authenticate: Digest realm="audiocodes.com",
nonce="11432d6bce58ddf02e3b5e1c77c010d2",
stale=FALSE,
algorithm=MD5
3.
According to the sub-header present in the WWW-Authenticate header, the correct
REGISTER request is formed.
4.
Since the algorithm is MD5, then:
5.
•
The username is equal to the endpoint phone number 122.
•
The realm return by the proxy is audiocodes.com.
•
The password from the ini file is AudioCodes.
•
The equation to be evaluated is (according to RFC this part is called A1)
‘122:audiocodes.com:AudioCodes’.
•
The MD5 algorithm is run on this equation and stored for future usage.
•
The result is ‘a8f17d4b41ab8dab6c95d3c14e34a9e1’.
Next, the par called A2 needs to be evaluated:
•
The method type is ‘REGISTER’.
•
Using SIP protocol ‘sip’.
•
Proxy IP from ini file is ‘10.2.2.222’.
•
The equation to be evaluated is ‘REGISTER:sip:10.2.2.222’.
•
The MD5 algorithm is run on this equation and stored for future usage.
•
The result is ’a9a031cfddcb10d91c8e7b4926086f7e’.
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7. IP Telephony Capabilities
Final stage:
•
The A1 result: The nonce from the proxy response is
‘11432d6bce58ddf02e3b5e1c77c010d2’.
•
The A2 result: The equation to be evaluated is
‘A1:11432d6bce58ddf02e3b5e1c77c010d2:A2’.
•
The MD5 algorithm is run on this equation. The outcome of the calculation is the
response needed by the device to register with the Proxy.
•
The response is ‘b9c45d0234a5abf5ddf5c704029b38cf’.
At this time, a new REGISTER request is issued with the following response:
REGISTER sip:10.2.2.222 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.200
From: <sip: [email protected]>;tag=1c23940
To: <sip: [email protected]>
Call-ID: [email protected]
Server: Audiocodes-Sip-Gateway/Mediant 2000/v.5.40.010.006
CSeq: 1 REGISTER
Contact: sip:[email protected]:
Expires:3600
Authorization: Digest, username: 122,
realm="audiocodes.com”,
nonce="11432d6bce58ddf02e3b5e1c77c010d2",
uri=”10.2.2.222”,
response=“b9c45d0234a5abf5ddf5c704029b38cf”
7.
Upon receiving this request and if accepted by the Proxy, the proxy returns a 200 OK
response closing the REGISTER transaction:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.200
From: <sip: [email protected]>;tag=1c23940
To: <sip: [email protected]>
Call-ID: [email protected]
Cseq: 1 REGISTER
Date: Thu, 26 Jul 2001 09:34:42 GMT
Server: Columbia-SIP-Server/1.17
Content-Length: 0
Contact: <sip:[email protected]>; expires="Thu, 26 Jul 2001 10:34:42
GMT"; action=proxy; q=1.00
Contact: <[email protected]:>; expires="Tue, 19 Jan 2038 03:14:07
GMT"; action=proxy; q=0.00
Expires: Thu, 26 Jul 2001 10:34:42 GMT
7.14.3 SIP Trunking between Enterprise and ITSPs
By implementing the device's enhanced and flexible routing configuration capabilities using
Proxy Sets, IP Groups, and Accounts, you can "design" complex routing schemes. This
section provides an example of an elaborate routing scheme for SIP trunking between an
Enterprise's PBX and two Internet Telephony Service Providers (ITSP), using AudioCodes'
device.
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Scenario: In this example, the Enterprise wishes to connect its TDM PBX to two different
ITSPs, by implementing a device in its network environment. It's main objective is for the
device to route Tel-to-IP calls to these ITSPs according to a dial plan. The device is to
register (on behalf of the PBX) to each ITSP, which implements two servers for redundancy
and load balancing. The Register messages are to use different URI's in the From, To, and
Contact headers per ITSP. In addition, all calls dialed from the Enterprise PBX with prefix
'02' is sent to the local PSTN. The figure below illustrates the example setup:
Figure 7-3: Example Setup for Routing Between ITSP and Enterprise PBX
¾ To configure call routing between Enterprise and two ITSPs using
the device, take these 8 steps:
1.
Enable the device to register to a Proxy / Registrar server, using the parameter
IsRegisterNeeded in the 'Proxy & Registration' page (refer to ''Proxy & Registration
Parameters'' on page 132).
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2.
7. IP Telephony Capabilities
In the 'Proxy Sets Table' page (refer to ''Proxy Sets Table'' on page 141), configure two
Proxy Sets and for each, enable Proxy Keep-Alive (using SIP OPTIONS) and 'round
robin' load-balancing method:
•
Proxy Set #1 includes two IP addresses of the first ITSP (ITSP 1) - 10.33.37.77
and 10.33.37.79 - and using UDP.
•
Proxy Set #2 includes two IP addresses of the second ITSP (ITSP 2) - 10.8.8.40
and 10.8.8.10 - and using TCP.
The figure below displays the configuration of Proxy Set ID #1. Perform similar
configuration for Proxy Set ID #2, but using different IP addresses.
Figure 7-4: Configuring Proxy Set ID #1 in the Proxy Sets Table Page
3.
In the 'IP Group Table' page (refer to ''Configuring the IP Groups'' on page 201),
configure the two IP Groups #1 and #2. Assign Proxy Sets #1 and #2 to IP Groups #1
and #2 respectively.
Figure 7-5: Configuring IP Groups #1 and #2 in the IP Group Table Page
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4.
In the 'Trunk Group Table' page (refer to “Configuring the Trunk Group Table” on page
195), enable the Trunks connected between the Enterprise's PBX and the device
(Trunk Group ID #1), and between the local PSTN and the device (Trunk Group ID
#2).
Figure 7-6: Assign the Trunk to Trunk Group ID #1 in the Trunk Group Table Page
5.
In the 'Trunk Group Settings' page (refer to ''Configuring the Trunk Group Settings'' on
page 197), configure 'Per Account' registration for Trunk Group ID #1 (without serving
IP Group).
Figure 7-7: Configuring Trunk Group #1 for Registration per Account in Trunk Group Settings
Page
6.
In the 'Account Table' page (refer to ''Configuring the Account Table'' on page 204),
configure the two Accounts for PBX trunk registration to ITSPs using the same Trunk
Group (i.e., ID #1), but different serving IP Groups #1 and #2. For each account, define
user name, password, and hostname, and ContactUser. The Register messages use
different URI's (Hostname and ContactUser) in the From, To, and Contact headers per
ITSP. Enable registration for both accounts. .
Figure 7-8: Configuring Accounts for PBX Registration to ITSPs in Account Table Page
7.
In the 'IP to Trunk Group Routing' page (refer to ''IP to Trunk Group Routing'' on page
181), configure IP-to-Tel routing for calls from ITSPs to Trunk Group ID #1 (see 1
below) and from the device to the local PSTN (see 2 below).
Figure 7-9: Configuring ITSP-to-Trunk Group #1 Routing in IP to Trunk Group Table Page
8.
In the 'Tel to IP Routing' page (refer to ''Tel to IP Routing Table'' on page 175),
configure Tel-to-IP routing rules for calls to ITSPs (see 1 below) and to local PSTN
(see 2 below) .
Figure 7-10: Configuring Tel-to-IP Routing to ITSPs in Tel to IP Routing Table Page
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7.15
7. IP Telephony Capabilities
Working with Supplementary Services
The device supports the following supplementary services:
„
Call Hold and Retrieve (refer to ''Call Hold and Retrieve'' on page 377).
„
Call Transfer (refer to ''Call Transfer'' on page 377).
„
Call Forward: when a callRerouting IE is received in a FACILITY message in response
to an outgoing SETUP message, the device sends a 3xx response to the IP side,
including the callRerouting destination number - only applicable to QSIG protocol
„
Call Waiting
The device SIP users are only required to enable the Hold and Transfer features. By
default, the Call Forward (supporting 30x redirecting responses) and Call Waiting (receipt of
182 response) features are enabled.
Notes:
•
All call participants must support the specific supplementary service that
is used.
•
When working with certain application servers (such as BroadSoft’s
BroadWorks) in client server mode (the application server controls all
supplementary services and keypad features by itself), the device's
supplementary services must be disabled.
7.15.1 Call Hold and Retrieve
Hold and Retrieve:
„
The party that initiates the hold is called the holding party; the other party is called the
held party. The device can't initiate Call Hold, but it can respond to hold requests and
as such, it's a help party.
„
After a successful Hold, the holding party hears a Dial tone (HELD_TONE defined in
the device's Call Progress Tones file).
„
After a successful retrieve, the voice is connected again.
„
The hold and retrieve functionalities are implemented by Re-INVITE messages. The IP
address 0.0.0.0 as the connection IP address or the string ‘a=inactive’ in the received
Re-INVITE SDP cause the device to enter Hold state and to play the Held tone
(configured in the device) to the PBX/PSTN. If the string ‘a=sendonly’ is received in the
SDP message, the device stops sending RTP packets, but continues to listen to the
incoming RTP packets. Usually, the remote party plays, in this scenario, Music on Hold
(MOH) and the device forwards the MOH to the held party.
You can also configure the device to keep a call on-hold for a user-defined time after which
the call is disconnected, using the ini file parameter HeldTimeout (refer to ''Supplementary
Services'' on page 159).
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7.15.2 Call Transfer
There are two types of call transfers:
„
Consultation Transfer:
The common way to perform a consultation transfer is as follows:
In the transfer scenario there are three parties: Party A = transferring, Party B =
transferred, Party C = transferred to.
•
A Calls B.
•
B answers.
•
A presses the hook-flash button and places B on-hold (party B hears a hold tone).
•
A dials C.
•
After A completes dialing C, A can perform the transfer by on-hooking the A
phone.
•
After the transfer is complete, B and C parties are engaged in a call.
The transfer can be initiated at any of the following stages of the call between A and C:
„
•
Just after completing dialing C phone number - transfer from setup.
•
While hearing Ringback – transfer from alert.
•
While speaking to C - transfer from active.
Blind Transfer:
Blind transfer is performed after we have a call between A and B, and party A decides
to immediately transfer the call to C without speaking with C. The result of the transfer
is a call between B and C (just like consultation transfer only skipping the consultation
stage).
Note: The device doesn't initiate call transfer, it only responds to call transfer
requests.
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8. Networking Capabilities
8
Networking Capabilities
8.1
Ethernet Interface Configuration
The device's Ethernet connection can be configured (using the ini file parameter
EthernetPhyConfiguration) for one of the following modes:
„
„
Manual mode:
•
10Base-T Half-Duplex or 10Base-T Full-Duplex
•
100Base-TX Half-Duplex or 100Base-TX Full-Duplex
Auto-Negotiation: chooses common transmission parameters such as speed and
duplex mode
The Ethernet connection should be configured according to the following recommended
guidelines:
„
When the device's Ethernet port is configured for Auto-Negotiation, the opposite port
must also operate in Auto-Negotiation. Auto-Negotiation falls back to Half-Duplex
mode when the opposite port is not in Auto-Negotiation mode, but the speed (i.e.,
10/100Base-T or 1000Base-TX) in this mode is always configured correctly.
Configuring the device to Auto-Negotiation mode while the opposite port is set
manually to Full-Duplex is invalid as it causes the device to fall back to Half-Duplex
mode while the opposite port is Full-Duplex. Any mismatch configuration can yield
unexpected functioning of the Ethernet connection.
„
When configuring the device's Ethernet port manually, the same mode (i.e., Half
Duplex or Full Duplex) and speed must be configured on the remote Ethernet port. In
addition, when the device's Ethernet port is configured manually, it is invalid to set the
remote port to Auto-Negotiation. Any mismatch configuration can yield unexpected
functioning of the Ethernet connection.
„
It's recommended to configure the port for best performance and highest bandwidth
(i.e., Full Duplex with 100Base-TX), but at the same time adhering to the guidelines
listed above.
Note that when remote configuration is performed, the device should be in the correct
Ethernet setting prior to the time this parameter takes effect. When, for example, the device
is configured using BootP/TFTP, the device performs many Ethernet-based transactions
prior to reading the ini file containing this device configuration parameter. To resolve this
problem, the device always uses the last Ethernet setup mode configured. In this way, if
you want to configure the device to operate in a new network environment in which the
current Ethernet setting of the device is invalid, you should first modify this parameter in the
current network so that the new setting holds next time the device is restarted. After
reconfiguration has completed, connect the device to the new network and restart it. As a
result, the remote configuration process that occurs in the new network uses a valid
Ethernet configuration.
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8.2
Ethernet Interface Redundancy
The device supports an Ethernet redundancy scheme. At the beginning of the start-up
procedure, the device tests whether the ‘primary’ Ethernet interface is connected, by
checking the existence of the Ethernet link carrier. If it's connected, the start-up procedure
commences as usual. If not, the start-up application tries the ‘secondary’ Ethernet interface.
If this interface is connected, the whole start-up procedure is performed using it. If both
interfaces are not connected, the start-up procedure commences using the parameters,
tables, and software residing on the device's non-volatile memory. Note that Ethernet
switchover occurs only once during the start-up procedure (at the beginning). If the Ethernet
interface fails after the selection is made, the device does not switch over to the second
port.
After start-up is complete and the operational software is running, the device continues to
use the Ethernet port used for software upload. The device switches over from one
Ethernet port to the other each time an Ethernet link carrier-loss is detected on the active
Ethernet port, and if the Ethernet link of the other port is operational. Switchover occurs
only once per link loss (i.e., the ‘secondary’ interface stays the active one even if the
‘primary’ interface has returned to life). After start-up, the device generates a gratuitous
ARP message each time a switchover occurs.
For correct functionality of the redundancy mechanism, it's recommended to configure both
links to the same mode. It is essential that both link partners (primary and secondary) have
the same capabilities. This ensures that whenever a switchover occurs, the device is able
to provide at least the same Ethernet services as were provided prior to the switchover. In
addition, it's recommended to set the physical secondary link prior to resetting the device
(since the MAC configuration cannot be changed thereafter).
Note that since the two Ethernet ports use the same MAC address, the external switches
connected to the device can in some cases create a noticeable switchover delay due to
their internal switching logic, though at the device level, the switchover delay is minimal
(milliseconds).
8.3
NAT (Network Address Translation) Support
Network Address Translation (NAT) is a mechanism that maps a set of internal IP
addresses used within a private network to global IP addresses, providing transparent
routing to end hosts. The primary advantages of NAT include (1) Reduction in the number
of global IP addresses required in a private network (global IP addresses are only used to
connect to the Internet); (2) Better network security by hiding its internal architecture.
The following figure illustrates the device's supported NAT architecture.
Figure 8-1: NAT Architecture
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8. Networking Capabilities
The design of SIP creates a problem for VoIP traffic to pass through NAT. SIP uses IP
addresses and port numbers in its message body and the NAT server can’t modify SIP
messages and therefore, can’t change local to global addresses. Two different streams
traverse through NAT: signaling and media. A device (located behind a NAT) that initiates a
signaling path has problems in receiving incoming signaling responses (they are blocked by
the NAT server). Furthermore, the initiating device must notify the receiving device where to
send the media.
To resolve these issues, the following mechanisms are available:
„
STUN (refer to ''STUN'' on page 381)
„
First Incoming Packet Mechanism (refer to ''First Incoming Packet Mechanism'' on
page 382)
„
RTP No-Op packets according to the avt-rtp-noop draft (refer to ''No-Op Packets'' on
page 382)
For information on SNMP NAT traversal, refer to the Product Reference Manual.
8.3.1
STUN
Simple Traversal of UDP through NATs (STUN), based on RFC 3489 is a client / server
protocol that solves most of the NAT traversal problems. The STUN server operates in the
public Internet and the STUN clients are embedded in end-devices (located behind NAT).
STUN is used both for the signaling and the media streams. STUN works with many
existing NAT types and does not require any special behavior.
STUN enables the device to discover the presence (and types) of NATs and firewalls
located between it and the public Internet. It provides the device with the capability to
determine the public IP address and port allocated to it by the NAT. This information is later
embedded in outgoing SIP / SDP messages and enables remote SIP user agents to reach
the device. It also discovers the binding lifetime of the NAT (the refresh rate necessary to
keep NAT ‘Pinholes’ open).
On startup, the device sends a STUN Binding Request. The information received in the
STUN Binding Response (IP address:port) is used for SIP signaling. This information is
updated every user-defined period (NATBindingDefaultTimeout).
At the beginning of each call and if STUN is required (i.e., not an internal NAT call), the
media ports of the call are mapped. The call is delayed until the STUN Binding Response
(that includes a global IP:port) for each media (RTP, RTCP and T.38) is received.
To enable STUN, perform the following:
„
Enable the STUN feature using either the Web interface (refer to ''Configuring the
Application Settings'' on page 57) or the ini file (set EnableSTUN to 1).
„
Define the STUN server address using one of the following methods:
„
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•
Define the IP address of the primary and the secondary (optional) STUN servers
using either the Web interface (refer to ''Configuring the Application Settings'' on
page 57) or the ini file (STUNServerPrimaryIP and STUNServerSecondaryIP). If
the primary STUN server isn’t available, the device attempts to communicate with
the secondary server.
•
Define the domain name of the STUN server using the ini file parameter
StunServerDomainName. The STUN client retrieves all STUN servers with an
SRV query to resolve this domain name to an IP address and port, sort the server
list, and use the servers according to the sorted list.
Use the ini file parameter NATBindingDefaultTimeout to define the default NAT binding
lifetime in seconds. STUN is used to refresh the binding information after this time
expires.
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8.3.2
•
STUN only applies to UDP (doesn’t support TCP and TLS).
•
STUN can’t be used when the device is located behind a symmetric NAT.
•
Use either the STUN server IP address (STUNServerPrimaryIP) or
domain name (STUNServerDomainName) method, with priority to the
first one.
First Incoming Packet Mechanism
If the remote device resides behind a NAT device, it’s possible that the device can activate
the RTP/RTCP/T.38 streams to an invalid IP address / UDP port. To avoid such cases, the
device automatically compares the source address of the incoming RTP/RTCP/T.38 stream
with the IP address and UDP port of the remote device. If the two are not identical, the
transmitter modifies the sending address to correspond with the address of the incoming
stream. The RTP, RTCP and T.38 can thus have independent destination IP addresses and
UDP ports.
You can disable the NAT mechanism by setting the ini file parameter DisableNAT to 1. The
two parameters EnableIpAddrTranslation and EnableUdpPortTranslation allow you to
specify the type of compare operation that occurs on the first incoming packet. To compare
only the IP address, set EnableIpAddrTranslation to 1, and EnableUdpPortTranslation to 0.
In this case, if the first incoming packet arrives with only a difference in the UDP port, the
sending addresses won’t change. If both the IP address and UDP port need to be
compared, then both parameters need to be set to 1.
8.3.3
No-Op Packets
The device's No-Op packet support can be used to verify Real-Time Transport Protocol
(RTP) and T.38 connectivity, and to keep NAT bindings and Firewall pinholes open. The
No-Op packets are available for sending in RTP and T.38 formats.
You can control the activation of No-Op packets by using the ini file parameter
NoOpEnable. If No-Op packet transmission is activated, you can control the time interval in
which No-Op packets are sent in the case of silence (i.e., no RTP or T.38 traffic). This is
performed using the ini file parameter NoOpInterval. For a description of the RTP No-Op ini
file parameters, refer to ''Networking Parameters'' on page 260.
„
RTP No-Op: The RTP No-Op support complies with IETF’s draft-wing-avt-rtp-noop03.txt (titled ‘A No-Op Payload Format for RTP’). This IETF document defines a No-Op
payload format for RTP. The draft defines the RTP payload type as dynamic. You can
control the payload type with which the No-Op packets are sent. This is performed
using the RTPNoOpPayloadType ini parameter (refer to ''Networking Parameters'' on
page 260). AudioCodes’ default payload type is 120.
„
T.38 No-Op: T.38 No-Op packets are sent only while a T.38 session is activated. Sent
packets are a duplication of the previously sent frame (including duplication of the
sequence number).
Note: Receipt of No-Op packets is always supported.
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8.4
8. Networking Capabilities
IP Multicasting
The device supports IP Multicasting level 1 according to RFC 2236 (i.e., IGMP version 2)
for RTP channels. The device is capable of transmitting and receiving Multicast packets.
8.5
Robust Reception of RTP Streams
This mechanism filters out unwanted RTP streams that are sent to the same port number
on the device. These multiple RTP streams can result from traces of previous calls, call
control errors, and deliberate attacks. When more than one RTP stream reaches the device
on the same port number, the device accepts only one of the RTP streams and rejects the
rest of the streams.
The RTP stream is selected according to the following: The first packet arriving on a newly
opened channel sets the source IP address and UDP port from which further packets are
received. Thus, the source IP address and UDP port identify the currently accepted stream.
If a new packet arrives whose source IP address or UDP port are different to the currently
accepted RTP stream, one of the following occurs:
8.6
„
The device reverts to the new RTP stream when the new packet has a source IP
address and UDP port that are the same as the remote IP address and UDP port that
were stated during the opening of the channel.
„
The packet is dropped when the new packet has any other source IP address and
UDP port.
Multiple Routers Support
Multiple routers support is designed to assist the device when it operates in a multiple
routers network. The device learns the network topology by responding to Internet Control
Message Protocol (ICMP) redirections and caches them as routing rules (with expiration
time).
When a set of routers operating within the same subnet serve as devices to that network
and intercommunicate using a dynamic routing protocol, the routers can determine the
shortest path to a certain destination and signal the remote host the existence of the better
route. Using multiple router support, the device can utilize these router messages to change
its next hop and establish the best path.
Note: Multiple Routers support is an integral feature that doesn’t require
configuration.
8.7
Simple Network Time Protocol Support
The Simple Network Time Protocol (SNTP) client functionality generates requests and
reacts to the resulting responses using the NTP version 3 protocol definitions (according to
RFC 1305). Through these requests and responses, the NTP client synchronizes the
system time to a time source within the network, thereby eliminating any potential issues
should the local system clock 'drift' during operation. By synchronizing time to a network
time source, traffic handling, maintenance, and debugging become simplified for the
network administrator.
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The NTP client follows a simple process in managing system time: the NTP client requests
an NTP update, receives an NTP response, and then updates the local system clock based
on a configured NTP server within the network.
The client requests a time update from a specified NTP server at a specified update
interval. In most situations, this update interval is every 24 hours based on when the system
was restarted. The NTP server identity (as an IP address) and the update interval are userdefined using either the Web interface (refer to ''Configuring the Application Settings'' on
page 57), the ini file (NTPServerIP and NTPUpdateInterval respectively), or an SNMP MIB
object (refer to the Product Reference Manual).
When the client receives a response to its request from the identified NTP server, it must be
interpreted based on time zone or location offset that the system is to a standard point of
reference called the Universal Time Coordinate (UTC). The time offset that the NTP client
uses is configurable using the Web interface (refer to ''Configuring the Application Settings''
on page 57), the ini file (NTPServerUTCOffset), or via an SNMP MIB object (refer to the
Product Reference Manual).
If required, the clock update is performed by the client as the final step of the update
process. The update is performed in such a way as to be transparent to the end users. For
instance, the response of the server may indicate that the clock is running too fast on the
client. The client slowly robs bits from the clock counter to update the clock to the correct
time. If the clock is running too slow, then in an effort to catch the clock up, bits are added
to the counter, causing the clock to update quicker and catch up to the correct time. The
advantage of this method is that it does not introduce any disparity in the system time that
is noticeable to an end user or that could corrupt call timeouts and timestamps.
8.8
IP QoS via Differentiated Services (DiffServ)
DiffServ is an architecture providing different types or levels of service for IP traffic. DiffServ
(according to RFC 2474) offers the capability to prioritize certain traffic types depending on
their priority, thereby, accomplishing a higher-level QoS at the expense of other traffic
types. By prioritizing packets, DiffServ routers can minimize transmission delays for timesensitive packets such as VoIP packets.
The device can be configured to set a different DiffServ value to IP packets according to
their class-of-service: Network, Premium Media, Premium Control, Gold, and Bronze. The
DiffServ parameters are described in ''Networking Parameters'' on page 260.
For the mapping of an application to its class-of-service, refer to ''IEEE 802.1p/Q (VLANs
and Priority)'' on page 385.
8.9
VLANS and Multiple IPs
8.9.1
Multiple IPs
Media, Control, and Management (OAMP) traffic in the device can be assigned one of the
following IP addressing schemes:
„
Single IP address for all traffic (i.e., for Media, Control, and OAMP).
„
Separate IP address for each of the three traffic types: The different traffic types
are separated into three dedicated networks. Instead of a single IP address, the device
is assigned three IP addresses and subnet masks, each relating to a different traffic
type. This architecture enables you to integrate the device into a three-network
environment that is focused on security and segregation. Each entity in the device
(e.g., Web and RTP) is mapped to a single traffic type (according to the table in ''IEEE
802.1p/Q (VLANs and Priority)'' on page 385) in which it operates.
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„
8. Networking Capabilities
Dual IP mode: The device is assigned two IP addresses for the different traffic types.
One IP address is assigned to a combination of two traffic types (Media and Control,
OAMP and Control, or OAMP and Media), while the other IP address is assigned to
whichever traffic type not included in this combination. For example, a typical scenario
using this mode includes one IP address assigned to Control and OAMP, and another
IP address assigned to Media.
For detailed information on integrating the device into a VLAN and multiple IPs network,
refer to ''Getting Started with VLANS and Multiple IPs'' on page 387. For detailed
information on configuring the multiple IP parameters, refer to ''Networking Parameters'' on
page 260.
Notes:
8.9.2
•
A default Gateway is supported only for the Media traffic type; for Control
and OAM traffic, use the 'IP Routing' table (refer to ''Configuring the IP
Routing Table'' on page 62).
•
The IP address and subnet mask used in the Single IP Network mode are
used for the OAM traffic type in the Multiple IP Network mode.
IEEE 802.1p/Q (VLANs and Priority)
The Virtual Local Area Network (VLAN) mechanism enables the device to be integrated into
a VLAN-aware environment that includes switches, routers and endpoints. When in VLANenabled mode, each packet is tagged with values that specify its priority (class-of-service /
IEEE 802.1p) and the identifier (traffic type) of the VLAN to which it belongs (Media,
Control, or OAMP / IEEE 802.1Q).
The class-of-service (CoS) mechanism can be utilized to accomplish Ethernet Quality of
Service (QoS). Packets sent by the device to the Ethernet network are divided into five
different-priority classes (Network, Premium Media, Premium Control, Gold, and Bronze).
The priority of each class is determined by a corresponding ini file parameter.
Traffic type tagging can be used to implement Layer 2 VLAN security. By discriminating
traffic into separate and independent domains, the information is preserved within the
VLAN. Incoming packets received from an incorrect VLAN are discarded.
The traffic tagging mechanism is as follows:
„
Outgoing packets (from the device to the switch): All outgoing packets are tagged,
each according to its interface (Control, Media or OAMP). If the device’s native VLAN
ID is identical to one of the other IDs (usually to the OAMP's VLAN ID), this ID (e.g.,
OAMP) is set to zero on outgoing packets (VlanSendNonTaggedOnNative set to 0).
This method is called Priority Tagging (p tag without Q tag). If the parameter
VlanSendNonTaggedOnNative is set to 1, the device sends regular packets (with no
VLAN tag).
„
Incoming packets (from the switch to the device): The switch sends all packets
intended for the device (according to the switch’s configuration) to the device without
altering them. For packets whose VLAN ID is identical to the switch’s PVID, the switch
removes the tag and sends a packet. The device accepts only packets that have a
VLAN ID identical to one of its interfaces (Control, Media or OAMP). Packets with a
VLAN ID that is 0 or untagged packets are accepted only if the device’s native VLAN
ID is identical to the VLAN ID of one of its interfaces. In this case, the packets are sent
to the relevant interface. All other packets are rejected.
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Media traffic type is assigned ‘Premium media’ CoS, Management traffic type is assigned
‘Bronze’ CoS, and Control traffic type is assigned ‘Premium control’ CoS. For example,
RTP/RTCP traffic is assigned the Media VLAN ID and ‘Premium media’ CoS, whereas Web
traffic is assigned the Management VLAN ID and ‘Bronze’ CoS. Each of these parameters
can be configured with a 802.1p/Q value: traffic type to VLAN ID, and CoS to 802.1p
priority.
Figure 8-2: Multiple Network Interfaces and VLANs
Notes:
•
For security, the VLAN mechanism is activated only when the device is
loaded from the flash memory. Therefore, when using BootP:
Load an ini file with VlanMode set to 1 and SaveConfiguration set to 1.
Then (after the device is active) reset the device with TFTP disabled or
by using any method except for BootP.
•
For information on how to configure VLAN parameters, refer to
''Configuring the IP Settings'' on page 50.
•
•
The device must be connected to a VLAN-aware switch and the switch’s
PVID must be equal to the device’s native VLAN ID.
The mapping of an application to its CoS and traffic type is shown in the table below:
Table 8-1: Traffic / Network Types and Priority
Application
Traffic / Network Types
Class-of-Service (Priority)
Debugging interface
Management
Bronze
Telnet
Management
Bronze
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8. Networking Capabilities
Application
Traffic / Network Types
Class-of-Service (Priority)
DHCP
Management
Network
Web server (HTTP)
Management
Bronze
SNMP GET/SET
Management
Bronze
Web server (HTTPS)
Management
Bronze
IPSec IKE
Determined by the service
Determined by the service
RTP traffic
Media
Premium media
RTCP traffic
Media
Premium media
T.38 traffic
Media
Premium media
SIP
Control
Premium control
SIP over TLS (SIPS)
Control
Premium control
Syslog
Management
Bronze
ICMP
Management
Determined by the initiator of the
request
ARP listener
Determined by the initiator of the
request
Network
SNMP Traps
Management
Bronze
DNS client
DNS (EnableDNSasOAM)
Network
Depends on traffic type:
NTP (EnableNTPasOAM)
NTP
ƒ
Control: Premium control
ƒ
Management: Bronze
NFS
NFSServers_VlanType in the
NFSServers table
8.9.3
Getting Started with VLANS and Multiple IPs
Gold
By default, the device operates without VLANs and multiple IPs, using a single IP address,
subnet mask and default Gateway IP address. This section provides an example of the
configuration required to integrate the device into a multiple IPs network withVLANs, using
the Web interface (refer to ''Integrating Using the Web Interface'' on page 388) and ini file
(refer to ''Integrating Using the ini File'' on page 390). The following table shows an example
configuration used in this subsection:
Table 8-2: Example of VLAN and Multiple IPs Configuration
Network
Type
IP Address
Subnet
Mask
Default Gateway
IP Address
VLAN ID
External Routing
Rule
OAMP
10.31.174.50
255.255.0.0
0.0.0.0
4
83.4.87.X
Control
10.32.174.50
255.255.0.0
0.0.0.0
5
130.33.4.6
Media
10.33.174.50
255.255.0.0
10.33.0.1
6
--
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Notes:
8.9.3.1
•
The values provided in this section are only used as an example.
•
Since a default Gateway is available only for the Media network, for the
device to be able to communicate with an external device/network on its
OAMP and Control networks, IP routing rules must be used.
Integrating Using the Web Interface
The procedure below describes how to integrate the device into a multiple IPs network
withVLANs, using the Web interface.
¾ To integrate the device into a multiple IPs network withVLANs
using the Web interface, take these 6 steps:
1.
Access the Web interface (refer to ''Accessing the Web Interface'' on page 20).
2.
Use the Software Upgrade Wizard (refer to ''Software Upgrade Wizard'' on page 236)
to load and burn the firmware version to the device (VLANs and multiple IPs support is
available only when the firmware is burned to flash).
3.
Configure the VLAN parameters by completing the following steps:
a.
Open the 'IP Settings' page (refer to ''Configuring the IP Settings'' on page 50).
b.
Modify the VLAN parameters to correspond to the values shown in the following
figure:
Figure 8-3: VLAN Configuration in the IP Settings Page
c.
4.
Click the Submit button to save your changes.
Configure the multiple IP parameters by completing the following steps:
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a.
8. Networking Capabilities
In the ‘IP Settings’ page, modify the IP parameters to correspond to the values
shown in the figure below. Note that the OAM, Control, and Media Network
Settings parameters appear only after you select the options ‘Multiple IP
Networks’ or 'Dual IP' in the field ‘IP Networking Mode’.
Figure 8-4: OAM, Control, Media IP Configuration in the IP Settings Page
Instead of configuring in the ‘IP Settings’ page, you can use the 'Multiple Interface
Table' page, which is accessed from the ‘IP Settings’ page by clicking the right-arrow
button alongside the label 'Multiple Interface Table' (refer to ''Configuring the
Multiple Interface Table'' on page 53). The 'Multiple Interface Table' page provides
greater configuration flexibility whereby you can also assign VLANs to the different
interfaces.
Figure 8-5: Multiple Interface Table Page
b.
Click the Submit button to save your changes.
Note: Configure the OAM parameters only if the OAM networking parameters are
different from the networking parameters used in the Single IP Network mode.
5.
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Configure the 'IP Routing' table to define static routing rules for the OAMP and Control
networks, since a default gateway isn’t supported on these networks:
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Open the ‘IP Routing Table’ page (refer to ''Configuring the IP Routing Table'' on
page 62).
a.
Figure 8-6: Static Routes for OAM/Control in IP Routing Table
b.
Use the Add New Entry to add the routing rules listed in the following table:
Destination IP Address
Destination Mask
Gateway IP Address
Hop Count
Interface
87.66.15.8
255.255.255.255
10.13.0.1
20
Control
85.44.115.50
255.255.255.0
10.31.0.1
20
OAMP
6.
8.9.3.2
Save your changes to flash memory (refer to ''Saving Configuration'' on page 230) and
reset the device (refer to ''Resetting the Device'' on page 228).
Integrating Using the ini File
The procedure below describes how to integrate the device into a multiple IPs network with
VLANs, using the ini file. The procedure below is based on the example setup described in
''Getting Started with VLANS and Multiple IPs'' on page 387.
¾ To integrate the device into a multiple IPs network withVLANs
using the ini file, take these 3 steps:
1.
Prepare an ini file (using the ini file table parameter InterfaceTable) with relevant
parameters:
•
If the BootP/TFTP utility and the OAMP interface are located on the same
network, the Native VLAN ID (VlanNativeVlanId) must be equal to the OAMP
VLAN ID (VlanOamVlanId), which in turn must be equal to the PVID of the switch
port to which the device is connected. Therefore, set the PVID of the switch port
to 4 (in this example).
•
Configure the OAMP parameters only if the OAMP networking parameters are
different from the networking parameters used in the Single IP Network mode.
•
The 'IP Routing' table is required to define static routing rules for the OAMP and
Control networks since a default Gateway isn’t supported for these networks.
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Below is an example of an ini file containing VLAN and Multiple IPs parameters:
; Interface Table Configuration:
[InterfaceTable]
FORMAT InterfaceTable_Index = InterfaceTable_ApplicationTypes,
InterfaceTable_IPv6InterfaceMode, InterfaceTable_IPAddress,
InterfaceTable_PrefixLength, InterfaceTable_Gateway,
InterfaceTable_VlanID, InterfaceTable_InterfaceName;
InterfaceTable 0 = 0, 0, 10.31.174.50, 16, 0.0.0.0, 4, OAMP;
InterfaceTable 0 = 1, 0, 10.33.174.50, 16, 10.33.0.1, 6, Media;
InterfaceTable 0 = 2, 0, 10.32.174.50, 16, 0.0.0.0, 5, Control;
[\InterfaceTable]
; VLAN related parameters:
VlanMode = 1
VlanNativeVlanId=4
; Routing Table Configuration:
; IP Routing table parameters
RoutingTableDestinationsColumn = 130.33.4.6, 83.4.87.6
RoutingTableDestinationMasksColumn = 255.255.255.255 ,
255.255.255.0
RoutingTableGatewaysColumn = 10.32.0.1 , 10.31.0.1
RoutingTableInterfacesColumn = 2,0
RoutingTableHopsCountColumn = 20,20
; Class Of Service parameters:
VlanNetworkServiceClassPriority = 7
VlanPremiumServiceClassMediaPriority = 6
VlanPremiumServiceClassControlPriority = 6
VlanGoldServiceClassPriority = 4
VlanBronzeServiceClassPriority = 2
NetworkServiceClassDiffServ = 48
PremiumServiceClassMediaDiffServ = 46
PremiumServiceClassControlDiffServ = 40
GoldServiceClassDiffServ = 26
BronzeServiceClassDiffServ = 10
; Application Type for applications:
EnableDNSasOAM = 1
EnableSCTPasControl = 1
EnableNTPasOAM = 1
2.
Use the BootP/TFTP utility (refer to the Product Reference Manual) to load and burn
the firmware version and the ini file you prepared in the previous step to the device
(multiple IPs and VLANs support is available only when the firmware is burned to
flash).
3.
Reset the device after disabling it on the BootP/TFTP utility.
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Instead of using the ini file table parameter InterfaceTable, you can configure multiple IPs
and VLANs using the individual ini file parameters, as shown below:
; VLAN Configuration
VlanMode=1
VlanOamVlanId=4
VlanNativeVlanId=4
VlanControlVlanId=5
VlanMediaVlanID=6
; Multiple IPs Configuration
EnableMultipleIPs=1
LocalMediaIPAddress=10.33.174.50
LocalMediaSubnetMask=255.255.0.0
LocalMediaDefaultGW=10.33.0.1
LocalControlIPAddress=10.32.174.50
LocalControlSubnetMask=255.255.0.0
LocalControlDefaultGW=0.0.0.0
LocalOAMPAddress=10.31.174.50
LocalOAMSubnetMask=255.255.0.0
LocalOAMDefaultGW=0.0.0.0
; IP Routing table parameters
RoutingTableDestinationsColumn = 130.33.4.6, 83.4.87.6
RoutingTableDestinationMasksColumn = 255.255.255.255,
255.255.255.0
RoutingTableGatewaysColumn = 10.32.0.1 , 10.31.0.1
RoutingTableInterfacesColumn = 1 , 0
RoutingTableHopsCountColumn = 20,20
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9. Advanced PSTN Configuration
Advanced PSTN Configuration
This section discusses advanced PSTN configurations.
9.1
Clock Settings
In a traditional TDM service network such as PSTN, both ends of the TDM connection must
be synchronized. If synchronization is not achieved, voice frames are either dropped (to
prevent a buffer overflow condition) or inserted (to prevent an underflow condition). In both
cases, connection quality and reliability is affected.
The device's clock settings can be configured to one of the following:
„
Generate its own timing signals
„
Use an internal clock
„
Recover a clock from one of the PSTN E1/T1 trunks
¾ To use the device's internal clock source, configure the following
parameters:
„
TDMBusClockSource = 1
„
ClockMaster = 1 (for all trunks)
¾ To use the recovered clock option, configure the following
parameters:
„
TDMBusClockSource = 4
„
ClockMaster_x = 0 (for all ‘slave’ trunks connected to PBX#1)
„
ClockMaster_x = 1 (for all ‘master’ trunks connected to PBX#2)
The above assumes that the device recovers its internal clock from one of the ‘slave’ trunks
connected to PBX#1 and provides clock to PBX#2 on its ‘master’ trunks. In addition, it's
necessary to define from which of the ‘slave’ trunks the device recovers its clock:
„
TDMBusPSTNAutoClockEnable = 1 (device automatically selects one of the
connected ‘slave’ trunks)
- Or -
„
TDMBusLocalReference = # (trunk number, where 0 is the first trunk - and the default)
Notes:
Version 5.6
•
To configure the TDM Bus Clock Source parameters using the Web
interface, refer to ''Configuring the TDM Bus Settings'' on page 218.
•
When the device is used in a ‘non-span’ configuration, the internal device
clock must be used (as explained above).
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9.2
Release Reason Mapping
This section describes the available mapping mechanisms of SIP responses to Q.850
Release Causes and vice versa. The existing mapping of ISDN Release Causes to SIP
Responses is described in ''Fixed Mapping of ISDN Release Reason to SIP Response'' on
page 394 and ''Fixed Mapping of SIP Response to ISDN Release Reason'' on page 396. To
override this hard-coded mapping and flexibly map SIP responses to ISDN Release
Causes, use the ini file (CauseMapISDN2SIP and CauseMapSIP2ISDN, as described in
''ISDN and CAS Interworking-Related Parameters'' on page 307) or the Web interface (refer
to ''Release Cause Mapping'' on page 189).
It is also possible to map the less commonly used SIP responses to a single default ISDN
Release Cause. Use the parameter DefaultCauseMapISDN2IP (described in ''ISDN and
CAS Interworking-Related Parameters'' on page 307) to define a default ISDN Cause that is
always used except when the following Release Causes are received: Normal Call Clearing
(16), User Busy (17), No User Responding (18) or No Answer from User (19). This
mechanism is only available for Tel-to-IP calls.
9.2.1
Reason Header
The device supports the Reason header according to RFC 3326. The Reason header
conveys information describing the disconnection cause of a call:
„
Sending Reason header: If a call is disconnected from the Tel side (ISDN), the
Reason header is set to the received Q.850 cause in the appropriate message (BYE /
CANCEL / final failure response) and sent to the SIP side. If the call is disconnected
because of a SIP reason, the Reason header is set to the appropriate SIP response.
„
Receiving Reason header: If a call is disconnected from the IP side and the SIP
message includes the Reason header, it is sent to the Tel side according to the
following logic:
•
If the Reason header includes a Q.850 cause, it is sent as is.
•
If the Reason header includes a SIP response:
•
9.2.2
♦
If the message is a final response, the response status code is translated to
Q.850 format and passed to ISDN.
♦
If the message isn’t a final response, it is translated to a Q.850 cause.
When the Reason header is received twice (i.e., SIP Reason and Q.850), the
Q.850 takes precedence over the SIP reason and is sent to the Tel side.
Fixed Mapping of ISDN Release Reason to SIP Response
The following table describes the mapping of ISDN release reason to SIP response.
Table 9-1: Mapping of ISDN Release Reason to SIP Response
ISDN
Release
Reason
SIP
Response
Description
Description
1
Unallocated number
404
Not found
2
No route to network
404
Not found
3
No route to destination
404
Not found
6
Channel unacceptable
406*
Not acceptable
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ISDN
Release
Reason
Description
SIP
Response
7
Call awarded and being delivered in an
established channel
500
16
Normal call clearing
17
User busy
486
Busy here
18
No user responding
408
Request timeout
19
No answer from the user
480
Temporarily unavailable
21
Call rejected
403
Forbidden
22
Number changed w/o diagnostic
410
Gone
26
Non-selected user clearing
404
Not found
27
Destination out of order
502
Bad gateway
28
Address incomplete
484
Address incomplete
29
Facility rejected
501
Not implemented
30
Response to status enquiry
501*
Not implemented
31
Normal unspecified
480
Temporarily unavailable
34
No circuit available
503
Service unavailable
38
Network out of order
503
Service unavailable
41
Temporary failure
503
Service unavailable
42
Switching equipment congestion
503
Service unavailable
43
Access information discarded
502*
Bad gateway
44
Requested channel not available
503*
Service unavailable
47
Resource unavailable
503
Service unavailable
49
QoS unavailable
503*
Service unavailable
50
Facility not subscribed
503*
Service unavailable
55
Incoming calls barred within CUG
403
Forbidden
57
Bearer capability not authorized
403
Forbidden
58
Bearer capability not presently available
503
Service unavailable
63
Service/option not available
503*
Service unavailable
65
Bearer capability not implemented
501
Not implemented
66
Channel type not implemented
480*
Temporarily unavailable
69
Requested facility not implemented
503*
Service unavailable
70
Only restricted digital information bearer
capability is available
503*
Service unavailable
79
Service or option not implemented
501
Not implemented
81
Invalid call reference value
502*
Bad gateway
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ISDN
Release
Reason
SIP
Response
Description
Description
82
Identified channel does not exist
502*
Bad gateway
83
Suspended call exists, but this call
identity does not
503*
Service unavailable
84
Call identity in use
503*
Service unavailable
85
No call suspended
503*
Service unavailable
86
Call having the requested call identity
has been cleared
408*
Request timeout
87
User not member of CUG
503
Service unavailable
88
Incompatible destination
503
Service unavailable
91
Invalid transit network selection
502*
Bad gateway
95
Invalid message
503
Service unavailable
96
Mandatory information element is
missing
409*
Conflict
97
Message type non-existent or not
implemented
480*
Temporarily not available
98
Message not compatible with call state
or message type non-existent or not
implemented
409*
Conflict
99
Information element non-existent or not
implemented
480*
Not found
100
Invalid information elements contents
501*
Not implemented
101
Message not compatible with call state
503*
Service unavailable
102
Recovery of timer expiry
408
Request timeout
111
Protocol error
500
Server internal error
127
Interworking unspecified
500
Server internal error
* Messages and responses were created because the ‘ISUP to SIP Mapping’ draft doesn’t
specify their cause code mapping.
9.2.3
Fixed Mapping of SIP Response to ISDN Release Reason
The following table describes the mapping of SIP response to ISDN release reason.
Table 9-2: Mapping of SIP Response to ISDN Release Reason
SIP
Response
Description
ISDN Release
Reason
Description
400*
Bad request
31
Normal, unspecified
401
Unauthorized
21
Call rejected
402
Payment required
21
Call rejected
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Response
9. Advanced PSTN Configuration
Description
ISDN Release
Reason
Description
403
Forbidden
21
Call rejected
404
Not found
1
Unallocated number
405
Method not allowed
63
Service/option unavailable
406
Not acceptable
79
Service/option not implemented
407
Proxy authentication
required
21
Call rejected
408
Request timeout
102
Recovery on timer expiry
409
Conflict
41
Temporary failure
410
Gone
22
Number changed w/o diagnostic
411
Length required
127
Interworking
413
Request entity too long
127
Interworking
414
Request URI too long
127
Interworking
415
Unsupported media type
79
Service/option not implemented
420
Bad extension
127
Interworking
480
Temporarily unavailable
18
No user responding
481*
Call leg/transaction doesn’t
exist
127
Interworking
482*
Loop detected
127
Interworking
483
Too many hops
127
Interworking
484
Address incomplete
28
Invalid number format
485
Ambiguous
1
Unallocated number
486
Busy here
17
User busy
488
Not acceptable here
31
Normal, unspecified
500
Server internal error
41
Temporary failure
501
Not implemented
38
Network out of order
502
Bad gateway
38
Network out of order
503
Service unavailable
41
Temporary failure
504
Server timeout
102
Recovery on timer expiry
505*
Version not supported
127
Interworking
600
Busy everywhere
17
User busy
603
Decline
21
Call rejected
604
Does not exist anywhere
1
Unallocated number
606*
Not acceptable
38
Network out of order
* Messages and responses were created because the ‘ISUP to SIP Mapping’ draft doesn’t
specify their cause code mapping.
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9.3
ISDN Overlap Dialing
Overlap dialing is a dialing scheme used by several ISDN variants to send and / or receive
called number digits one after the other (or several at a time). This is in contrast to en-bloc
dialing in which a complete number is sent.
The device can optionally support ISDN overlap dialing for incoming ISDN calls for the
entire device by setting the ini file parameter ISDNRxOverlap to 1, or per E1/T1 span by
setting ISDNRxOverlap_x to 1 (where x represents the number of the trunk). For
configuring ISDN overlap dialing using the Web interface, refer to ''Configuring the Trunk
Settings'' on page 82.
To play a Dial tone to the ISDN user side when an empty called number is received, set
ISDNINCallsBehavior = 65536 (bit #16). This results in the Progress Indicator to be
included in the SetupAck ISDN message.
The device stops collecting digits (for ISDN-to-IP calls) when:
„
The sending device transmits a ‘sending complete’ IE in the ISDN Setup or the
following INFO messages to signal that no more digits are going to be sent.
„
The inter-digit timeout (configured by the parameter TimeBetweenDigits) expires. The
default for this timeout is 4 seconds.
„
The maximum allowed number of digits (configured by the parameter MaxDigits) is
reached. The default is 30 digits.
„
A match is found with the defined digit map (configured by the parameter,
DigitMapping).
Relevant parameters (described in ''PSTN Parameters'' on page 303):
9.4
„
ISDNRxOverlap
„
ISDNRxOverlap_x
„
TimeBetweenDigits
„
MaxDigits
„
ISDNInCallsBehavior
„
DigitMapping
ISDN Non-Facility Associated Signaling (NFAS)
In regular T1 ISDN trunks, a single 64 kbps channel carries signaling for the other 23 Bchannels of that particular T1 trunk. This channel is called the D-channel and usually
resides on timeslot # 24. The ISDN Non-Facility Associated Signaling (NFAS) feature
enables the use of a single D-channel to control multiple PRI interfaces.
With NFAS it is possible to define a group of T1 trunks, called an NFAS group, in which a
single D-channel carries ISDN signaling messages for the entire group. The NFAS group’s
B-channels are used to carry traffic such as voice or data. The NFAS mechanism also
enables definition of a backup D-channel on a different T1 trunk, to be used if the primary
D-channel fails.
The NFAS group can comprise up to 10 T1 trunks. Each T1 trunk is called an ‘NFAS
member’. The T1 trunk whose D-channel is used for signaling is called the ‘Primary NFAS
Trunk’. The T1 trunk whose D-channel is used for backup signaling is called the ‘Backup
NFAS Trunk’. The primary and backup trunks each carry 23 B-channels while all other
NFAS trunks each carry 24 B-channels.
The device supports up to 9 NFAS groups. Each group must contain different T1 trunks.
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The NFAS group is identified by an NFAS GroupID number (possible values are 1 to 9). To
assign a number of T1 trunks to the same NFAS group, use the ini file parameter
NFASGroupNumber_x = groupID (where x is the physical trunk ID (0 to the maximum
number of trunks) or the Web interface (refer to ''Configuring the Trunk Settings'' on page
82).
The parameter ‘DchConfig_x = Trunk_type’ defines the type of NFAS trunk. Trunk_type is
set to 0 for the primary trunk, to 1 for the backup trunk, and to 2 for an ordinary NFAS trunk.
‘x’ depicts the physical trunk ID (0 to the maximum number of trunks). You can also use the
Web interface (refer to ''Configuring the Trunk Settings'' on page 82).
For example, to assign the first four T1 trunks to NFAS group #1, in which trunk #0 is the
primary trunk and trunk #1 is the backup trunk, use the following configuration:
NFASGroupNumber 0
NFASGroupNumber_1
NFASGroupNumber_2
NFASGroupNumber_3
DchConfig_0 = 0
DchConfig_1 = 1
DchConfig_2 = 2
DchConfig_3 = 2
=
=
=
=
1
1
1
1
;Primary T1 trunk
;Backup T1 trunk
;24 B-channel NFAS trunk
;24 B-channel NFAS trunk
The NFAS parameters are described in ''PSTN Parameters'' on page 303.
9.4.1
NFAS Interface ID
Several ISDN switches require an additional configuration parameter per T1 trunk that is
called ‘Interface Identifier’. In NFAS T1 trunks, the Interface Identifier is sent explicitly in
Q.931 Setup / Channel Identification IE for all NFAS trunks, except for the B-channels of
the Primary trunk (refer to note below).
The Interface ID can be defined per member (T1 trunk) of the NFAS group, and must be
coordinated with the configuration of the Switch. The default value of the Interface ID is
identical to the number of the physical T1 trunk (0 for the first trunk, 1 for the second T1
trunk, and so on, up to the maximum number of trunks).
To define an explicit Interface ID for a T1 trunk (that is different from the default), use the
following parameters:
„
ISDNIBehavior_x = 512 (x = 0 to the maximum number of trunks identifying the
device's physical trunk)
„
ISDNNFASInterfaceID_x = ID (x = 0 to 255)
Notes:
Version 5.6
•
Usually the Interface Identifier is included in the Q.931 Setup/Channel
Identification IE only on T1 trunks that doesn’t contain the D-channel.
Calls initiated on B-channels of the Primary T1 trunk, by default, don’t
contain the Interface Identifier. Setting the parameter ISDNIBehavior_x to
2048’ forces the inclusion of the Channel Identifier parameter also for the
Primary trunk.
•
The parameter ISDNNFASInterfaceID_x = ID can define the ‘Interface ID’
for any Primary T1 trunk, even if the T1 trunk is not a part of an NFAS
group. However, to include the Interface Identifier in Q.931
Setup/Channel Identification IE configure ISDNIBehavior_x = 2048 in the
ini file.
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9.4.2
Working with DMS-100 Switches
The DMS-100 switch requires the following NFAS Interface ID definitions:
„
InterfaceID #0 for the Primary trunk
„
InterfaceID #1 for the Backup trunk
„
InterfaceID #2 for a 24 B-channel T1 trunk
„
InterfaceID #3 for a 24 B-channel T1 trunk, and so on for subsequent T1 trunks
For example, if four T1 trunks on a device are configured as a single NFAS group with
Primary and Backup T1 trunks that is used with a DMS-100 switch, the following
parameters should be used:
NFASGroupNumber 0
NFASGroupNumber_1
NFASGroupNumber_2
NFASGroupNumber_3
DchConfig_0 = 0
DchConfig_1 = 1
DchConfig_2 = 2
DchConfig_3 = 2
= 1
= 1
= 1
= 1
;Primary T1 trunk
;Backup T1 trunk
;B-Channel NFAS trunk
;B-channel NFAS trunk
If there is no NFAS Backup trunk, the following configuration should be used:
ISDNNFASInterfaceID 0 = 0
ISDNNFASInterfaceID_1 = 2
ISDNNFASInterfaceID_2 = 3
ISDNNFASInterfaceID_3 = 4
ISDNIBehavior = 512
;This parameter should be added because of
;ISDNNFASInterfaceID coniguration above
NFASGroupNumber_0 = 1
NFASGroupNumber_1 = 1
NFASGroupNumber_2 = 1
NFASGroupNumber_3 = 1
DchConfig_0 = 0
;Primary T1 trunk
DchConfig_1 = 2
;B-Channel NFAS trunk
DchConfig_2 = 2
;B-Channel NFAS trunk
DchConfig_3 = 2
;B-channel NFAS trunk
9.4.3
Creating an NFAS-Related Trunk Configuration
The procedures for creating and deleting an NFAS group must be performed in the correct
order, as described below.
¾ To create an NFAS Group, take these 3 steps:
1.
If there’s a backup (‘secondary’) trunk for this group, it must be configured first.
2.
Configure the primary trunk before configuring any NFAS (‘slave’) trunk.
3.
Configure NFAS (‘slave’) trunks.
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¾ To stop / delete an NFAS Group, take these 3 steps:
1.
Stop or delete (by setting ProtocolType to 0, i.e., 'None') all NFAS (‘slave’) trunks.
2.
Stop or delete (by setting ProtocolType to 0, i.e., 'None') the backup trunk if a backup
trunk exists.
3.
Stop or delete (by setting ProtocolType to 0, i.e., 'None') the primary trunk.
Notes:
9.5
•
All trunks in the group must be configured with the same values for trunk
parameters TerminationSide, ProtocolType, FramingMethod, and
LineCode.
•
After stopping or deleting the backup trunk, delete the group and then
reconfigure it.
•
NFAS groups cannot be configured on-the-fly.
Redirect Number and Calling Name (Display)
The following tables define the device's redirect number and calling name (Display) support
for various PRI variants according to NT (Network Termination) / TE (Termination
Equipment) interface direction:
Table 9-3: Calling Name (Display)
NT/TE Interface
DMS-100
NI-2
4/5ESS
Euro ISDN
QSIG
NT-to-TE
Yes
Yes
Yes
Yes
Yes
TE-to-NT
Yes
Yes
Yes
No
Yes
Table 9-4: Redirect Number
NT/TE Interface
DMS-100
NI-2
4/5ESS
Euro ISDN
QSIG
NT-to-TE
Yes
Yes
Yes
Yes
Yes
TE-to-NT
Yes
Yes
Yes
Yes*
Yes
* When using ETSI DivertingLegInformation2 in a Facility IE (not Redirecting Number IE).
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9.6
Automatic Gain Control (AGC)
Automatic Gain Control (AGC) adjusts the energy of the output signal to a required level
(i.e., volume). This feature compensates for near-far gain differences. AGC estimates the
energy of the incoming signal (from the IP or PSTN, determined by the parameter
AGCRedirection), calculates the essential gain, and then performs amplification. Feedback
ensures that the output signal is not clipped. You can define the required Gain Slope in
decibels per second (using the parameter AGCGainSlop) and the required signal energy
threshold (using the parameter AGCTargetEnergy).
When the AGC first detects an incoming signal, it begins operating in Fast Mode, which
allows the AGC to adapt quickly when a conversation starts. This means that the Gain
Slope is 8 dB/sec for the first 1.5 seconds. After this period, the Gain Slope is changed to
the user-defined value. You can disable or enable the AGC's Fast Mode feature, using the
ini file parameter AGCDisableFastAdaptation. After Fast Mode is used, the signal should be
off for two minutes in order to have the feature turned on again.
To configure AGC, refer to ''Configuring the IPmedia Settings'' on page 76.
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10. Tunneling Applications
Tunneling Applications
This section discusses TDM and QISG tunneling, supported by the device.
10.1
TDM Tunneling
The device's TDM Tunneling feature allows you to tunnel groups of digital trunk spans or
timeslots (B-channels) over the IP network. TDM Tunneling utilizes the device's internal
routing (without Proxy control) capabilities to receive voice and data streams from TDM
(E1/T1/J1/) spans or individual timeslots, convert them into packets, and then transmit them
over the IP network (using point-to-point or point-to-multipoint device distributions). A
device opposite it (or several devices when point-to-multipoint distribution is used) converts
the IP packets back into TDM traffic. Each timeslot can be targeted to any other timeslot
within a trunk in the opposite device.
When TDM Tunneling is enabled ('Enable TDM Tunneling' parameter is set to 'Enable' on
the originating device -- refer to ''Configuring the Digital Gateway Parameters'' on page
207), the originating device automatically initiates SIP calls from all enabled B-channels
belonging to the E1/T1/J1 spans that are configured with the protocol type ‘Transparent’ (for
ISDN trunks) or ‘Raw CAS’ (for CAS trunks). The called number of each call is the internal
phone number of the B-channel from where the call originates. The 'IP to Trunk Group
Routing' table (refer to ''IP to Trunk Group Routing'' on page 181) is used to define the
destination IP address of the terminating device. The terminating device automatically
answers these calls if its E1/T1 protocol type is set to ‘Transparent’ (ProtocolType = 5) or
‘Raw CAS’ (ProtocolType = 3 for T1 and 9 for E1) and the parameter ChannelSelectMode
is set to 0 (By Phone Number).
Note: It's possible to configure both devices to also operate in symmetric mode. To
do so, set EnableTDMOverIP to 1 and configure the 'Tel to IP Routing' tables
in both devices. In this mode, each device (after it's reset) initiates calls to the
second device. The first call for each B-channel is answered by the second
device.
The device continuously monitors the established connections. If for some reason, one or
more calls are released, the device automatically re-establishes these ‘broken’ connections.
In addition, when a failure in a physical trunk or in the IP network occurs, the device reestablishes the tunneling connections when the network is restored.
Note: It's recommended to use the keep-alive mechanism for each connection, by
activating the ‘session expires’ timeout and using Re-INVITE messages.
By utilizing the ‘Profiles’ mechanism (refer to ''Configuring the Profile Definitions'' on page
190), you can configure the TDM Tunneling feature to choose different settings based on a
timeslot or groups of timeslots. For example, you can use low-bit-rate vocoders to transport
voice and ‘Transparent’ coder to transport data (e.g., for D-channel). You can also use
Profiles to assign ToS (for DiffServ) per source -- a timeslot carrying data or signaling is
assigned a higher priority value than a timeslot carrying voice.
For tunneling of E1/T1 CAS trunks, set the protocol type to 'Raw CAS' (ProtocolType = 3 /
9) and enable RFC 2833 CAS relay mode ('CAS Transport Type' parameter is set to 'CAS
RFC2833 Relay' -- refer to ''Configuring the Voice Settings'' on page 66).
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Note: For TDM over IP, the 'Caller ID Transport Type' parameter must be set to
'Disable', i.e., transparent (refer to ''Configuring the Fax / Modem / CID
Settings'' on page 67).
Below is an example of ini files for two devices implementing TDM Tunneling for four E1
spans. Note that in this example both devices are dedicated to TDM tunneling.
Terminating Side:
EnableTDMOverIP = 1
;E1_TRANSPARENT_31
ProtocolType_0 = 5
ProtocolType_1 = 5
ProtocolType_2 = 5
ProtocolType_3 = 5
[PREFIX]
FORMAT PREFIX_Index = PREFIX_DestinationPrefix,
PREFIX_DestAddress, PREFIX_SourcePrefix, PREFIX_ProfileId,
PREFIX_MeteringCode, PREFIX_DestPort;
Prefix 1 = '*,10.8.24.12';
[\PREFIX]
;IP address of the device in the opposite
;location
;Channel selection by Phone number.
ChannelSelectMode = 0
;Profiles can be used do define different coders per B-channels
;such as Transparent
;coder for B-channels (timeslot 16) that carries PRI ;signaling.
[TrunkGroup]
FORMAT TrunkGroup_Index = TrunkGroup_TrunkGroupNum,
TrunkGroup_FirstTrunkId, TrunkGroup_LastTrunkId,
TrunkGroup_FirstBChannel, TrunkGroup_LastBChannel,
TrunkGroup_FirstPhoneNumber, TrunkGroup_ProfileId,
TrunkGroup_Module;
TrunkGroup 1 = 0,0,0,1,31,1000,1;
TrunkGroup 1 = 0,1,1,1,31,2000,1;
TrunkGroup 1 = 0,2,2,1,31,3000,1;
TrunkGroup 1 = 0,3,3,1,31,4000,1;
TrunkGroup 1 = 0,0,0,16,16,7000,2;
TrunkGroup 1 = 0,1,1,16,16,7001,2;
TrunkGroup 1 = 0,2,2,16,16,7002,2;
TrunkGroup 1 = 0,3,3,16,16,7003,2;
[/TrunkGroup]
[CoderName]
FORMAT CoderName_Index = CoderName_Type, CoderName_PacketInterval,
CoderName_rate, CoderName_PayloadType, CoderName_Sce;
CoderName 0 = 'g7231';
CoderName 1 = 'Transparent';
CoderName 5 = 'g7231';
CoderName 6 = 'Transparent';
[/CoderName]
[TelProfile]
FORMAT TelProfile_Index = TelProfile_ProfileName,
TelProfile_TelPreference, TelProfile_CodersGroupID,
TelProfile_IsFaxUsed, TelProfile_JitterBufMinDelay,
TelProfile_JitterBufOptFactor, TelProfile_IPDiffServ,
TelProfile_SigIPDiffServ, TelProfile_DtmfVolume,
TelProfile_InputGain, TelProfile_VoiceVolume,
TelProfile_EnableReversePolarity,
TelProfile_EnableCurrentDisconnect,
TelProfile_EnableDigitDelivery, TelProfile_EnableEC,
TelProfile_MWIAnalog, TelProfile_MWIDisplay,
TelProfile_FlashHookPeriod, TelProfile_EnableEarlyMedia,
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TelProfile ProgressIndicator2IP;
TelProfile 1 = voice,$$,1,$$,$$,$$,$$,$$,$$,$$;
TelProfile 2 = data,$$,2,$$,$$,$$,$$,$$,$$,$$;
[\TelProfile]
Originating Side:
;E1 TRANSPARENT 31
ProtocolType_0 = 5
ProtocolType_1 = 5
ProtocolType_2 = 5
ProtocolType_3 = 5
;Channel selection by Phone number.
ChannelSelectMode = 0
[TrunkGroup]
FORMAT TrunkGroup_Index = TrunkGroup_TrunkGroupNum,
TrunkGroup_FirstTrunkId, TrunkGroup_LastTrunkId,
TrunkGroup_FirstBChannel, TrunkGroup_LastBChannel,
TrunkGroup_FirstPhoneNumber, TrunkGroup_ProfileId,
TrunkGroup_Module;
TrunkGroup 0 = 0,0,0,1,31,1000,1;
TrunkGroup 0 = 0,1,1,1,31,2000,1;
TrunkGroup 0 = 0,2,2,1,31,3000,1;
TrunkGroup 0 = 0,3,1,31,4000,1;
TrunkGroup 0 = 0,0,0,16,16,7000,2;
TrunkGroup 0 = 0,1,1,16,16,7001,2;
TrunkGroup 0 = 0,2,2,16,16,7002,2;
TrunkGroup 0 = 0,3,3,16,16,7003,2;
[\TrunkGroup]
[CoderName]
FORMAT CoderName_Index = CoderName_Type, CoderName_PacketInterval,
CoderName_rate, CoderName_PayloadType, CoderName_Sce;
CoderName 1 = 'g7231';
CoderName 2 = 'Transparent';
[\CoderName]
[TelProfile]
FORMAT TelProfile_Index = TelProfile_ProfileName,
TelProfile_TelPreference, TelProfile_CodersGroupID,
TelProfile_IsFaxUsed, TelProfile_JitterBufMinDelay,
TelProfile_JitterBufOptFactor, TelProfile_IPDiffServ,
TelProfile_SigIPDiffServ, TelProfile_DtmfVolume,
TelProfile_InputGain, TelProfile_VoiceVolume,
TelProfile_EnableReversePolarity,
TelProfile_EnableCurrentDisconnect,
TelProfile_EnableDigitDelivery, TelProfile_EnableEC,
TelProfile_MWIAnalog, TelProfile_MWIDisplay,
TelProfile_FlashHookPeriod, TelProfile_EnableEarlyMedia,
TelProfile_ProgressIndicator2IP;
TelProfile_1 = voice,$$,1,$$,$$,$$,$$,$$,$$,$$
TelProfile_2 = data,$$,2,$$,$$,$$,$$,$$,$$,$$
[\TelProfile]
10.2
QSIG Tunneling
The device supports QSIG tunneling over SIP according to IETF draft 'Tunnelling of QSIG
over SIP' (draft-elwell-sipping-qsig-tunnel-03) and the ECMA-355/ISO/IEC 22535 standard.
This method enables all QSIG messages to be sent as raw data in corresponding SIP
messages using a dedicated message body. This mechanism is useful for two QSIG
subscribers (connected to the same or different QSIG PBX) to communicate with each
other over an IP network. Tunneling is supported in both directions (Tel-to-IP and IP-to-Tel).
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The term tunneling means that messages are transferred ‘as is’ to the remote side without
being converted (QSIGÆSIPÆQSIG). The advantage of tunneling over QSIG-to-SIP
interworking is that by using interworking, QSIG functionality can only be partially achieved.
When tunneling is used, all QSIG capabilities are supported, whereas the tunneling medium
(the SIP network) does not need to process these messages.
QSIG messages are transferred in SIP messages in a separate Multipurpose Internet Mail
Extensions (MIME) body. Therefore, if a message contains more than one body (e.g., SDP
and QSIG), multipart MIME must be used. The Content-Type of the QSIG tunneled
message is ‘application/QSIG’. In addition, the device adds a Content-Disposition header in
the following format:
Content-Disposition: signal; handling=required.
„
Call setup (originating device): The QSIG SETUP request is encapsulated in the
SIP INVITE message without being altered. After the SIP INVITE request is sent, the
device doesn’t encapsulate the subsequent QSIG message until a SIP 200 OK
response is received. If the originating device receives a 4xx, 5xx, or 6xx response, it
disconnects the QSIG call with a ‘no route to destination’ cause.
„
Call setup (terminating device): After the terminating device receives a SIP INVITE
request with a 'Content-Type: application/QSIG', it sends the encapsulated QSIG
SETUP message to the Tel side and sends a 200 OK response (no 1xx response is
sent) to IP. The 200 OK response includes an encapsulated QSIG CALL
PROCEEDING message (without waiting for a CALL PROCEEDING message from
the Tel side). If tunneling is disabled and the incoming INVITE includes a QSIG body,
a 415 response is sent.
„
Mid-call communication: After the SIP connection is established, all QSIG messages
are encapsulated in SIP INFO messages.
„
Call tear-down: The SIP connection is terminated once the QSIG call is complete.
The RELEASE COMPLETE message is encapsulated in the SIP BYE message that
terminates the session.
To enable QSIG tunneling, set the parameter EnableQSIGTunneling to 1 on both the
originating and terminating devices, and the parameter ISDNDuplicateQ931BuffMode to
128 (duplicate all messages) (both parameters are described in ''ISDN and CAS
Interworking-Related Parameters'' on page 307).
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11. Supplied SIP Software Package
Supplied SIP Software Package
The table below lists the standard SIP software package supplied with the SIP device.
Table 11-1: Supplied Software Package
File Name
Description
Ram.cmp file
Mediant_SIP_xxx.cmp
Image file containing the software for the Mediant 2000.
ini files
Mediant_SIP_T1.ini
Sample ini file for Mediant 2000 E1 device.
Mediant_SIP_E1.ini
Sample ini file for Mediant 2000 T1 device.
Usa_tones_xx.dat
Default loadable Call Progress Tones dat file
Usa_tones_xx.ini
Call Progress Tones ini file (used to create dat file)
voice_prompts.dat
Sample loadable Voice Prompts dat file
Utilities
DConvert
TrunkPack Downloadable Conversion Utility - to create Call Progress
Tones, Voice Prompts, and CAS files
ACSyslog
Syslog server
BootP
BootP/TFTP configuration utility
CAS Protocol Files
Used for various signaling types, such as E_M_WinkTable.dat
MIB Files
MIB library for SNMP browser
CAS Capture Tool
Utility that is used to convert CAS traces to textual form
ISDN Capture Tool
Utility that is used to convert ISDN traces to textual form
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12. Selected Technical Specifications
Selected Technical Specifications
The technical specifications of the Mediant 2000 is listed in the table below:
Note: All specifications in this document are subject to change without prior notice.
Table 12-1: Mediant 2000 Functional Specifications
Function
Specification
Trunk & Channel Capacity
Capacity with E1
1, 2, 4, 8 or 16 E1 spans, supporting channel capacity as follows:
ƒ
30 Channels on 1 E1 span with gateway-1 only
ƒ
60 Channels on 2 E1 spans with gateway-1 only
ƒ
120 Channels on 4 E1 spans with gateway-1 only
ƒ
240 Channels on 8 E1 spans with gateway-1 only
ƒ
480 Channels on 16 E1 spans with gateway-1 and gateway-2
Note: Channel capacity depends on configuration settings.
Capacity with T1
1, 2, 4, 8 or 16 T1 spans, supporting channel capacity as follows:
ƒ
24 Channels on 1 T1 span with gateway-1 only
ƒ
48 Channels on 2 T1 spans with gateway-1 only
ƒ
96 Channels on 4 T1 spans with gateway-1 only
ƒ
192 Channels on 8 T1 spans with gateway-1 only
ƒ
384 Channels on 16 T1 spans with gateway-1 and gateway-2
Note: Channel capacity depends on configuration settings.
Voice & Tone Characteristics
Voice Compression
G.711 PCM at 64 kbps µ-law/A-law; EG.711 µ-law/A-law at 64 kbps;
G.723.1 MP-MLQ at 5.3 or 6.3 kbps; G.726 at 32 kbps ADPCM; G.729
CS-ACELP 8 kbps Annex A / B; EVRC; AMR; Transparent; GSM Full
Rate; Microsoft GSM; iLBC; QCELP
Silence Suppression
ƒ
G.723.1 Annex A
ƒ
G.729 Annex B
ƒ
PCM and ADPCM: Standard Silence Descriptor (SID) with
Proprietary Voice Activity Detection (VAD) and Comfort Noise
Generation (CNG)
Packet Loss
Concealment
G.711 appendix 1; G.723.1; G.729 a/b
Echo Cancellation
G.165 and G.168 2000, configurable tail length per device from 32 to
128 msec
DTMF Detection and
Generation
Dynamic range 0 to -25 dBm, compliant with TIA 464B and Bellcore TRNWT-000506.
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Function
Specification
DTMF Transport (inband)
Mute, transfer in RTP payload or relay in compliance with RFC 2833
Answer Detector
Answer detection
Answer Machine
Detector
Detects whether voice or an answering machine is answering the call.
Note: When implementing Answer Machine Detector, channel capacity
may be reduced.
Call Progress Tone
Detection and
Generation
32 tones: single tone, dual tones or AM tones, programmable frequency
& amplitude; 64 frequencies in the range 300 to 1980 Hz, 1 to 4
cadences per tone, up to 4 sets of ON/OFF periods
Output Gain Control
-32 dB to +31 dB in steps of 1 dB
Input Gain Control
-32 dB to +31 dB in steps of 1 dB
Fax and Modem Transport Modes
Real time Fax Relay
ƒ
Group 3 real-time fax relay up to 14400 bps with automatic fallback
ƒ
Tolerant network delay (up to 9 seconds round trip delay)
ƒ
T.30 (PSTN) and T.38 (IP) compliant (real-time fax)
ƒ
CNG tone detection & Relay per T.38
ƒ
Answer tone (CED or AnsAm) detection & Relay per T.38
Fax Transparency
Automatic fax bypass (pass-through) to G.711, ADPCM or NSE bypass
mode
Modem Transparency
Automatic switching (pass-through) to PCM, ADPCM or NSE bypass
mode for modem signals (V.34 or V.90 modem detection)
Protocols
VoIP Signaling Protocol
SIP RFC 3261
Communication
Protocols
ƒ
RTP/RTCP packetization
ƒ
IP stack (UDP, TCP, RTP)
ƒ
Remote Software load (TFTP, HTTP and HTTPS)
ƒ
PRI (ETSI Euro ISDN, ANSI NI2, 4/5ESS, DMS 100, QSIG, Japan
INS1500, Australian Telecom, New Zealand Telecom, Hong Kong
Variant, Korean MIC)
ƒ
E1/T1 CAS protocols: MFC R2, E&M wink start
ƒ
Immediate start, delay start, loop start, ground start
ƒ
Feature Group B, D for E1/T1
ƒ
DTMF (TIA 464A)
ƒ
MF-R1, MFC R2
ƒ
User-defined Call Progress Tones
Telephony Protocols
In-Band Signaling
Interfaces
Telephony Interface
SIP User's Manual
1, 2, 4, 8 or 16 E1/T1/J1 Balanced 120/100 Ohm, or 75 Ohm using a
BNC to RJ-45 dual E1/T1 G.703 Balun adapter.
Note: The following Balun adaptors were tested and certified by
AudioCodes:
ƒ
Manufacture Name: AC&E (Part Number: B04040072)
ƒ
Manufacture Name: RIT (Part Number: R3712271)
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12. Selected Technical Specifications
Function
Specification
Network Interface
Two 10/100Base-TX, half or full duplex with auto-negotiation
RS-232 Interface
RS-232 terminal interface provided by DB-9 connector on rear panel
(available only on the 1, 2 and 4-span configurations)
LED Indicators
LED Indications on
Front Panel
Power, ACT/Fail, T1/E1 status, LAN status, Swap ready indication
Connectors & Switches
Rear Panel
Trunks 1 to 8 and 9 to
16
Two 50-pin female Telco connectors (DDK57AE-40500-21D) or 8 RJ48c connectors for trunks 1 to 8 only
Ethernet 1 and 2
Two 10/100Base-TX, RJ-45 shielded connectors
RS-232
DB-9 Console port
AC Power
ƒ
Standard IEC320 Appliance inlet
ƒ
Dual (fully redundant) power supply (optional)
ƒ
2-pin terminal block (screw connection type) suitable for field wiring
applications connecting DC Power connector MSTB2.5/2-STF (5.08
mm) from Phoenix Contact
ƒ
Bonding and earthing: 6-32-UNC screw is provided - correct ring
terminal and 16 AWG wire minimum must be used
ƒ
Or crimp connection (refer to note below)
DC Power
Note: To meet UL approval, customers must fulfill the criteria below:
2-pin terminal block (crimp connection type) comprising a Phoenix
Contact
ƒ
Adaptor: Shroud MSTBC2,5/2-STZF-5,08
ƒ
Contacts: MSTBC-MT0,5-1,0
ƒ
Cable: 18 AWG x 1.5 m length
ƒ
Universal 90 to 260 VAC 1A max, 47-63 Hz
ƒ
Dual redundant power supply (optional)
ƒ
1 or 2 span: 39.7 W
ƒ
4 spans: 42.1 W (approx.)
ƒ
8 spans: 45.3 W
Physical
AC Power Supply
AC Power Consumption
DC Power Supply
(optional)
36 to 72 VDC (nominal 48 VDC), 4A max, floating input
DC Power Consumption
ƒ
1 or 2 span: 28.8 W
ƒ
4 spans: 32.8 W
ƒ
8 spans: 36.4 W
ƒ
Operating Temp: 0 to 40°C (32 to 104°F)
ƒ
Short Term Operating Temp (per NEBS): 0 to 55°C (32 to 131°F)
ƒ
Storage: -40 to 70°C (-40 to 158°F)
ƒ
Humidity: 10 to 90% non-condensing
Environmental (DC)
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Function
Environmental (AC)
Hot Swap
Specification
ƒ
Operating Temp: 0 to 40°C (32 to 104°F)
ƒ
Storage: -40 to 70°C (-40 to 158°F)
ƒ
Humidity: 10 to 90% non-condensing
ƒ
cPCI blades are full hot-swappable
ƒ
Power supplies are redundant, but not hot-swappable
Enclosure Dimensions
445 x 44 x 300 mm (17.5 x 1.75 x 12 inches)
Weight
Approx. 4.8 kg fully populated (16 spans); 4.2 kg for 1 span
Installation
1U 19-inch 2-slot cPCI chassis; rack-, shelf-, or desktop-mount options.
Rack mount using two side brackets - 2 additional (rear) side brackets
optional
cPCI Blade
Control Processor
Motorola PowerQUICC 8260
Control Processor
Memory
SDRAM 64* - 128 MB (*on 60-channel models)
Signal Processors
AudioCodes AC486 VoIP DSP based on TI DSP TMS5541 – each core
at 133 MHz
PCI Bus Interface
33 MHz, 32 bit, slave mode (PICMG 2.0 revision 2.1)
Physical
6U single cPCI slot. PICMG 2.0, R2.1 and R2.16 and R.3.0
CompactPCI™ blade
Supply Voltages and
Power Consumption
(typical)
ƒ
480 channels: 40.7 W; 3 A at 5 V; 7.8 A at 3.3 V
ƒ
240 channels: 24 W; 1.5 A at 5 V; 5 A at 3.3 V
ƒ
120 channels: 18.4 W; 0.9 A at 5 V; 4.2 A at 3.3 V
Environmental
Humidity: 10 to 90% non-condensing
Cooling
ƒ
500 Linear Feet per Minute (LFM) at 50°C ambient temp. supporting
480 ports
ƒ
400 LFM at 50°C ambient temp. supporting 400 ports
ƒ
300 LFM at 50 °C ambient temp. supporting 240 ports
ƒ
E1/T1 status
ƒ
LAN status
ƒ
Status of device (Fail, ACT, Power, and Swap Ready)
Diagnostics
Front panel Status LEDs
Syslog events
Supported by Syslog Server, per RFC 3164 IETF standard.
SNMP MIBs and Traps
SNMP v2c; SNMP v3
Management
Configuration
Configuration of device using Web browser or ini files
Management and
Maintenance
ƒ
SNMP v2c; SNMP v3
ƒ
Syslog (RFC 3164)
ƒ
Web Management (via HTTP or HTTPS)
ƒ
Telnet
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Function
Specification
Type Approvals
Telecommunication
Standards
ƒ
IC CS03; FCC part 68
ƒ
Chassis and Host telecom card comply with IC CS03; FCC part 68;
CTR 4, CTR 12 & CTR 13; JATE; TS.016; TSO; Anatel, Mexico
Telecom, Russia CCC, ASIF S016, ASIF S038
Safety and EMC
Standards
ƒ
UL 60 950-1, FCC part 15 Class B, (Class A with SUN 2080 CPU
card)
ƒ
CE Mark: EN 55022 Class B (Class A with SUN 2080 CPU card), EN
60950-1, EN 55024, EN 300 386
ƒ
TS001
ƒ
NEBS Level 3: GR-63-Core, GR-1089-Core, Type 1 & 3. Approved
for DC powered version
ƒ
Complies with ETS 301019; ETS 300019-1, -2, -3. (T 1.1, T 2.3,
T3.2)
ƒ
Approved for AudioCodes or DC powered versions
Environmental
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13. Glossary
Glossary
Table 13-1: Glossary of Terms
Term
Meaning
ADPCM
Adaptive Differential PCM - voice compression
AIS
Alarm Indication Signal
A-law
Standard companding algorithm, used in European digital communications
systems to optimize the dynamic range of an analog signal for digitizing.
AMD
Answering Machine Detection
AOR
Address of Record
AWG
American Wire Gauge
bps
Bits per second
BootP
AudioCodes Proprietary Bootstrap Loader Utility
CAS
Channel Associated Signaling
CoS
Class of Service
CMP
Compressed File (device Firmware)
cPCI
Compact PCI (Industry Standard)
CPT
Call Progress Tones
dB
Decibels
DHCP
Dynamic Host Control Protocol
DID
Direct Inward Dial
DiffServ
Differentiated Services
DNS
Domain Name System (or Server)
DR
Debug Recording
DS1
1.544 Mbps USA Digital Transmission System (see E1 and T1)
DSP
Digital Signal Processor (or Processing)
DTMF
Dual Tone Multiple Frequency (Touch Tone)
E1
2.048 Mbps European Digital Transmission System (see T1)
ETSI
European Telecommunications Standards Institute
FQDN
Fully Qualified Domain Name
GRUU
Globally Routable User Agent URIs
ICMP
Internet Control Message Protocol
IE
Information Element (ISDN layer 3 protocol, basic building block)
IETF
Internet Engineering Task Force
IKE
Internet Key Exchange (for IPSec)
IP
Internet Protocol
IPSec
IP Security
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Term
Meaning
ISDN
Integrated Services Digital Network
ISO
International Standards Organization
ITU
International Telecommunications Union
ITU-T
Telecommunications section of the ITU
IVR
Interactive Voice Response
Jitter
Variation of interpacket timing interval
kbps
Kilobit per second. 1,000 bits per second
LAPD
Line Access Protocol for the D-channel
LFA
Loss of Frame Alignment
LOF
Loss of Frame
Mbps
Megabit per second. Million bits per second
MIB
Management Information Base
MLPP
Multilevel Precedence and Preemption
ms or msec
Millisecond; a thousandth part of a second
MSCML
Media Server Control Markup Language
NT
Network Termination (ISDN)
MWI
Message Waiting Indicator
NAPTR
Naming Authority Pointer
NAT
Network Address Translation
NFAS
Non-Facility Associated Signalling (ISDN PRI)
NFS
Network File System
NPI
Numbering Plan Indicator
NTP
Network Time Protocol
OAMP
Operations, Administration, Maintenance and Provisioning
OSI
Open Systems Interconnection (Industry Standard)
PBX
Private Branch Exchange
PCI
Personal Computer Interface (Industry Standard)
PCM
Pulse-Code Modulation
PI
Progress Indicator
PKI
Public-Key Infrastructures
POTS
Plain Old Telephone System or Service
PRT
Prerecorded Tones (File)
PRI
Primary Rate Interface (ISDN)
PSTN
Public Switched Telephone Network
PVID
Port VLAN ID (VLAN ID assignment to Ethernet packet by switch)
QoS
Quality of Service
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Term
Meaning
RAI
Remote Alarm Indication
RFC
Request for Comment issued by IETF
RTCP
Real-Time Transport (RTP) Control Protocol
RTP
Real-Time Transport Protocol
SA
Security Associations (contains encryption keys and profile used by IPSec to
encrypt the IP stream)
SAS
Stand Alone Survivability Feature
SDP
Session Description Protocol
SIP
Session Initiation Protocol
SMDI
Simplified Message Desk Interface
SME
Small and Medium-sized Enterprise
SNMP
Simple Network Management Protocol
SRTP
Secure Real-Time Transport Protocol
SRV
Service Record
SSH
Secure Shell
SSL
Secure Socket Layer (also known as Transport Layer Security (TLS))
STUN
Simple Traversal of UDP through NATs
T1
1.544 Mbps USA Digital Transmission System (see E1 and DS1)
TCP
Transmission Control Protocol
TCP/IP
Transmission Control Protocol / Internet Protocol
TE
Terminal Equipment (ISDN)
TDM
Time-Division Multiplexing
TFTP
Trivial File Transfer Protocol
TLS
Transport Layer Security
TON
Type of Numbering
UA
SIP User Agent
UDP
User Datagram Protocol
URI (SIP URIs)
SIP Uniform Resource Indicators
VBD
Voice-band data
VLAN
Virtual Local Area Network
VoIP
Voice over Internet Protocol
VoP
Voice over Packet(s)
VP
Voice Prompts (File)
VPN
Virtual Private Network
µ-Law
A companding algorithm, used in the digital telecommunication systems
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